JPH06112854A - Adaptive signal processing unit - Google Patents

Adaptive signal processing unit

Info

Publication number
JPH06112854A
JPH06112854A JP13035092A JP13035092A JPH06112854A JP H06112854 A JPH06112854 A JP H06112854A JP 13035092 A JP13035092 A JP 13035092A JP 13035092 A JP13035092 A JP 13035092A JP H06112854 A JPH06112854 A JP H06112854A
Authority
JP
Japan
Prior art keywords
processing
adaptive
coefficient data
filter
dsp
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Pending
Application number
JP13035092A
Other languages
Japanese (ja)
Inventor
Takashi Nakamoto
貴士 中本
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
NEC Corp
Original Assignee
NEC Corp
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by NEC Corp filed Critical NEC Corp
Priority to JP13035092A priority Critical patent/JPH06112854A/en
Publication of JPH06112854A publication Critical patent/JPH06112854A/en
Pending legal-status Critical Current

Links

Landscapes

  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
  • Noise Elimination (AREA)

Abstract

PURPOSE:To revise flexibly an algorithm for calculating a filter coefficient and to reduce the cost by improving a processing efficiency of the adaptive signal processing unit comprising a DSP. CONSTITUTION:A processing unit is provided with a DSP 1 implementing adaptive filter processing and RAMs 3, 4 storing filter coefficient data, and a DSP 2 reading the filter coefficient data from the RAMs 3, 4 and applying the adaptive filter processing to them, and the filter coefficient data are subject to correction processing.

Description

【発明の詳細な説明】Detailed Description of the Invention

【0001】[0001]

【産業上の利用分野】本発明は適応信号処理装置に関
し、特にエコーキャンセラやノイズキャンセラ等の適応
フィルタ処理に用いる適応信号処理装置に関する。
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an adaptive signal processing apparatus, and more particularly to an adaptive signal processing apparatus used for adaptive filter processing such as echo canceller and noise canceller.

【0002】[0002]

【従来の技術】近年、電話回線のエコーキャンセラやノ
イズキャンセラ等に、ディジタル信号処理装置(DS
P)を応用した適応信号処理装置が用いられてきてい
る。上記適応信号処理装置は、予測誤差の二乗平均値を
最小化するようにフィルタ係数の修正を行なう最小二乗
平均値(LMS)アルゴリズムに基ずく適応フィルタ処
理を行なう。
2. Description of the Related Art In recent years, digital signal processing devices (DS) have been used as echo cancellers and noise cancellers for telephone lines.
An adaptive signal processing device applying P) has been used. The adaptive signal processing device performs adaptive filter processing based on a least mean square (LMS) algorithm that corrects filter coefficients so as to minimize the mean square value of prediction errors.

【0003】以下に、上記適応フィルタ処理の概要を説
明する。
The outline of the adaptive filter processing will be described below.

【0004】まず、適応フィルタに対する現在の時刻T
0までの入力信号列ベクトルS(k)は次式により示さ
れる。
First, the current time T for the adaptive filter
The input signal sequence vector S (k) up to 0 is represented by the following equation.

【0005】 [0005]

【0006】また、フィルタ係数ベクトルA(k)は次
式により示される。
The filter coefficient vector A (k) is expressed by the following equation.

【0007】 [0007]

【0008】現在の時刻T0までの入力信号列による次
の時刻T1のサンプルの予測値P(k+1)は次式によ
り示される。
The predicted value P (k + 1) of the sample at the next time T1 according to the input signal train until the current time T0 is shown by the following equation.

【0009】 P(k+1)=A(k)・S(k)・・・・・・・・・・・・・・・・(3) また、時刻T1のサンプルに対するフィルタ係数の修正
式A(k+1)は次式により示される。
P (k + 1) = A (k) · S (k) (3) Further, the correction formula A (of the filter coefficient for the sample at time T1 k + 1) is shown by the following equation.

【0010】 A(k+1)=A(k)+K(k+1)・e(k)・・・・・・・・・(4) ここで、e(k)は誤差信号であり、ゲインKはLMS
アルゴリズムの場合次式で表される。
A (k + 1) = A (k) + K (k + 1) .e (k) ... (4) where e (k) is an error signal and the gain K is LMS.
In the case of an algorithm, it is expressed by the following equation.

【0011】 K(k+1)=S(k)・・・・・・・・・・・・・・・・・・・・・(5) 実用的には、伝送路誤りの影響を軽減するために、フィ
ルタ係数の修正式A(k+1)としては次式が用いられ
る。
K (k + 1) = S (k) (5) Practically, in order to reduce the influence of the transmission path error. In addition, the following expression is used as the correction expression A (k + 1) of the filter coefficient.

【0012】 A(k+1)=w・A(k)+K(k+1)・e(k) (0<w<1)・・・・・・・・・・・・・・・・(6) ここで、wは重み係数である。A (k + 1) = w · A (k) + K (k + 1) · e (k) (0 <w <1) (6) where Where w is a weighting factor.

【0013】図2に上記適応フィルタ処理のシグナルフ
ローを示す。
FIG. 2 shows a signal flow of the adaptive filter processing.

【0014】図2において、上記シグナルフローは乗算
処理M1〜M10と、入力信号に1サンプリング周期分
の時間の遅延を与える遅延処理D1〜D5と、加算処理
A1〜A5とを含んでいる。
In FIG. 2, the signal flow includes multiplication processes M1 to M10, delay processes D1 to D5 for delaying an input signal by a time corresponding to one sampling period, and addition processes A1 to A5.

【0015】ノイズキャンセラの場合には、入力信号S
(k)は、雑音だけの観測信号と次の時刻T1の入力サ
ンプルの予測値とに、P(k+1)は雑音の推定値に、
誤差信号e(k)は雑音混入状態の音声信号と雑音の推
定値との差にそれぞれ相当する。
In the case of a noise canceller, the input signal S
(K) is the noise-only observation signal and the predicted value of the input sample at the next time T1, and P (k + 1) is the noise estimated value.
The error signal e (k) corresponds to the difference between the voice signal in the noise-containing state and the estimated value of the noise.

【0016】上記適応フィルタ処理を行なう適応信号処
理装置は、一般に、ディジタル信号処理装置(DSP)
を用いて構成される。
An adaptive signal processing device for performing the adaptive filter processing is generally a digital signal processing device (DSP).
Is constructed using.

【0017】従来の適応信号処理装置は、適応フイルタ
を実現する処理と、フィルタ係数を修正する処理とを同
一のDSP内で行なっていた。通常の有限インパルス応
答(FIR)フィルタ処理用に特化しているDSPを用
いることにより、上記適応フイルタの実現処理は高効率
で達成できる。しかし、上記フィルタ係数の修正処理の
効率は高くはない。また、逆も同様であり、さらに、い
ずれかの処理にも特化していない汎用DSPの使用で
は、高効率の処理の実現は困難であった。一方、上記適
応信号処理専用にハードウェア構成を特化したDSPも
あるが、高価であり、また、係数算出のアルゴリズムを
フレキシブルに変更することが困難であるというもので
あった。
In the conventional adaptive signal processing apparatus, the processing for realizing the adaptive filter and the processing for correcting the filter coefficient are performed within the same DSP. By using a DSP specialized for normal finite impulse response (FIR) filter processing, the processing for realizing the adaptive filter can be achieved with high efficiency. However, the efficiency of the filter coefficient correction process is not high. The reverse is also true, and it is difficult to realize highly efficient processing by using a general-purpose DSP that is not specialized for any processing. On the other hand, there is a DSP having a specialized hardware configuration dedicated to the adaptive signal processing, but it is expensive and it is difficult to flexibly change the coefficient calculation algorithm.

【0018】[0018]

【発明が解決しようとする課題】上述した従来の適応信
号処理装置は、適応フイルタ実現処理と、フィルタ係数
修正処理とを同一のDSP内で行なっていたので、汎用
DSPでは全体の高効率の処理の実現は困難であり、い
ずれか一方の処理に特化したDSPの使用では、他方の
処理の効率向上が困難であるという欠点がった。また、
適応信号処理専用にハードウェア構成を特化したDSP
もあるが高価であり、また係数算出のアルゴリズムをフ
レキシブルに変更することが困難であるというという欠
点があった。
In the conventional adaptive signal processing apparatus described above, the adaptive filter realization processing and the filter coefficient correction processing are performed in the same DSP, so that the general-purpose DSP is highly efficient overall. However, there is a drawback that it is difficult to improve the efficiency of the other process by using the DSP specialized for one of the processes. Also,
DSP with specialized hardware configuration for adaptive signal processing
However, there are drawbacks that it is expensive and it is difficult to flexibly change the coefficient calculation algorithm.

【0019】[0019]

【課題を解決するための手段】本発明の適応信号処理装
置は、係数データを用いて適応フィルタ処理を行なう第
一の信号処理プロセッサと、前記係数データを格納し前
記第一の信号処理プロセッサに出力する係数データ記憶
手段と、前記係数データを修正処理しこの修正処理した
修正係数データを前記係数データ記憶手段に格納する第
二の信号処理プロセッサとを備えて構成されている。
An adaptive signal processing apparatus according to the present invention includes a first signal processor for performing adaptive filter processing using coefficient data, and a first signal processor for storing the coefficient data. It comprises a coefficient data storage means for outputting and a second signal processor for modifying the coefficient data and storing the modified coefficient data thus modified in the coefficient data storage means.

【0020】[0020]

【実施例】次に、本発明の実施例について図面を参照し
て説明する。
Embodiments of the present invention will now be described with reference to the drawings.

【0021】図1は本発明の適応信号処理装置の一実施
例を示すブロック図である。
FIG. 1 is a block diagram showing an embodiment of the adaptive signal processing apparatus of the present invention.

【0022】本実施例の適応信号処理装置は、図1に示
すように、適応フィルタ実現処理を行なうDSP1と、
フィルタ係数の修正処理を行なうDSP2と、フィルタ
係数を格納したRAM3と、修正フィルタ係数を格納し
たRAM4と、DSP1,2およびRAM3,4との間
のアドレスバスの切替を行うマルチプレクサMX1,M
X2と、DSP1,2およびRAM3,4との間のデー
タバスの切替を行うマルチプレクサMX3,MX4と、
サンプリング周期のタイミング信号FSで制御されるマ
ルチプレクサMX1〜MX4の切替制御用のトグルフリ
ップフロップF1と、インバータI1,I2とを備えて
構成されている。
As shown in FIG. 1, the adaptive signal processing apparatus of the present embodiment includes a DSP 1 for performing adaptive filter realization processing,
A DSP 2 for correcting the filter coefficient, a RAM 3 for storing the filter coefficient, a RAM 4 for storing the corrected filter coefficient, and multiplexers MX1, M for switching the address bus between the DSP 1, 2 and RAMs 3, 4.
Multiplexers MX3, MX4 for switching the data bus between X2 and the DSPs 1, 2 and RAMs 3, 4;
A toggle flip-flop F1 for switching control of the multiplexers MX1 to MX4 controlled by the timing signal FS of the sampling cycle, and inverters I1 and I2 are provided.

【0023】次に、本実施例の動作について説明する。Next, the operation of this embodiment will be described.

【0024】図2に示したLMSアルゴリズムによる適
応フィルタ処理のシグナルフローは、図3に示すよう
に、(A)のフィルタ実現処理フローと、(B)のフィ
ルタ係数修正処理のフローとに分解することができる。
The signal flow of the adaptive filter processing by the LMS algorithm shown in FIG. 2 is decomposed into a filter realization processing flow of (A) and a filter coefficient correction processing flow of (B) as shown in FIG. be able to.

【0025】フィルタ実現処理フローは、図3(A)に
示すようにFIRフィルタ処理であり、乗算処理M11
〜M21と、入力信号に1サンプリング周期分の時間の
遅延を与える遅延処理D11〜D15と、加算処理A1
1とを含んでいる。
The filter realization processing flow is FIR filter processing as shown in FIG. 3A, and multiplication processing M11
To M21, delay processing D11 to D15 for delaying the input signal by a time corresponding to one sampling cycle, and addition processing A1
Including 1 and.

【0026】フィルタ係数修正処理フローは、図3
(B)に示すように帰還型の処理であり、乗算処理M3
1〜M33と、入力信号に1サンプリング周期分の時間
の遅延を与える遅延処理D31〜D33と、加算処理A
31〜A33とを含んでいる。
The filter coefficient correction processing flow is shown in FIG.
As shown in (B), this is a feedback type process, and the multiplication process M3
1 to M33, delay processing D31 to D33 for delaying the input signal by a time corresponding to one sampling cycle, and addition processing A
31 to A33 are included.

【0027】また、a(n),b(n)は、それぞれ修
正対象のフィルタ係数およびフィルタ係数a(n)の修
正に必要な修正データをそれぞれ示す。
Further, a (n) and b (n) respectively indicate the filter coefficient to be corrected and the correction data necessary for correcting the filter coefficient a (n).

【0028】図1において、まず、DSP1は、RAM
3よりフィルタ係数データa(n)を読出し、適応フィ
ルタ実現処理を行なう。また、フィルタ係数データa
(n)の読出し後に修正データb(n)をRAM3に書
込む。一方、DSP2は、RAM4より修正データb
(n)を読出し、フィルタ係数データa(n)の修正値
を算出して結果をRAM4に書込む。1サンプリング周
期内での一連の処理が終了したならば、トグルフリップ
フロップF1,インバータI1,I2により、マルチプ
レクサMX1〜MX4を切替え、DSP1,2およびR
AM3,4との間のアドレスバスおよびデータバスの切
替を行う。この切替動作を、フィルタ係数データの修正
が必要な周期、すなわち、サンプリング周期FSで行な
うことによりRAM3,4を介してDSP1,2間のデ
ータ相互転送を可能としている。
In FIG. 1, first, the DSP 1 is a RAM
The filter coefficient data a (n) is read from 3 and adaptive filter realization processing is performed. Also, the filter coefficient data a
After reading (n), the correction data b (n) is written in the RAM 3. On the other hand, the DSP 2 stores the correction data b in the RAM 4
(N) is read, the correction value of the filter coefficient data a (n) is calculated, and the result is written in the RAM 4. When a series of processing within one sampling period is completed, the multiplexers MX1 to MX4 are switched by the toggle flip-flop F1, the inverters I1 and I2, and the DSPs 1, 2, and R are switched.
The address bus and the data bus are switched between AM3 and AM4. By performing this switching operation in a cycle in which the filter coefficient data needs to be corrected, that is, in the sampling cycle FS, mutual data transfer between the DSPs 1 and 2 is enabled via the RAMs 3 and 4.

【0029】以上、本発明の実施例を説明したが、本発
明は上記実施例に限られることなく種々の変形が可能で
ある。
Although the embodiments of the present invention have been described above, the present invention is not limited to the above embodiments, and various modifications can be made.

【0030】たとえば、フィルタ係数データと修正デー
タとをそれぞれ格納した2つのRAMを切替えて用いる
代りに、2ポートRAMを用い、係数修正周期に同期し
て2つのDSPのアクセス領域が重ならないように交互
に入れ代るアドレスを発生して動作させることも、本発
明の主旨を逸脱しない限り適用できることは勿論であ
る。
For example, instead of switching and using two RAMs respectively storing filter coefficient data and correction data, a 2-port RAM is used so that the access areas of two DSPs do not overlap in synchronization with the coefficient correction cycle. Needless to say, the generation and operation of alternating addresses can be applied without departing from the spirit of the present invention.

【0031】[0031]

【発明の効果】以上説明したように、本発明の適応信号
処理装置は、適応フィルタ処理を行なう第一の信号処理
プロセッサと、係数データを格納し上記第一の信号処理
プロセッサに出力する係数データ記憶手段と、上記係数
データを修正処理する第二の信号処理プロセッサとを備
えることにより、それぞれの処理に特化した信号処理プ
ロセッサを効率的に動作させることができるので、適応
信号処理全体の処理性能が向上するという効果がある。
As described above, the adaptive signal processing apparatus of the present invention includes a first signal processor for performing adaptive filter processing and coefficient data for storing coefficient data and outputting it to the first signal processor. By including the storage means and the second signal processing processor that corrects the coefficient data, the signal processing processor specialized for each processing can be efficiently operated. This has the effect of improving performance.

【0032】また、適応信号処理専用のDSPに比較し
安価であり、さらに係数算出のアルゴリズムをフレキシ
ブルに変更することも容易であるという効果がある。
Further, there is an effect that it is cheaper than a DSP dedicated to adaptive signal processing, and that the coefficient calculation algorithm can be easily changed flexibly.

【図面の簡単な説明】[Brief description of drawings]

【図1】本発明の適応信号処理装置の一実施例を示すブ
ロック図である。
FIG. 1 is a block diagram showing an embodiment of an adaptive signal processing device of the present invention.

【図2】適応フィルタ処理におけるシグナルフローを示
すシグナルフロー図である。
FIG. 2 is a signal flow diagram showing a signal flow in adaptive filter processing.

【図3】本実施例の適応信号処理装置の適応フィルタ処
理におけるシグナルフローを示すシグナルフロー図であ
る。
FIG. 3 is a signal flow diagram showing a signal flow in adaptive filter processing of the adaptive signal processing device of the present embodiment.

【符号の説明】[Explanation of symbols]

1,2 DSP 3,4 RAM F1 トグルフリップフロップ I1,I2 インバータ MX1〜MX4 マルチプレクサ A1〜A5,A11,A31〜A33 加算処理 D1〜D5,D11〜D15,D31〜D33 遅延
処理 M1〜M10,M11〜M21,M31〜M33 乗
算処理
1, 2 DSP 3, 4 RAM F1 toggle flip-flop I1, I2 inverter MX1 to MX4 multiplexer A1 to A5, A11, A31 to A33 addition processing D1 to D5, D11 to D15, D31 to D33 delay processing M1 to M10, M11 to M21, M31 to M33 Multiplication processing

Claims (1)

【特許請求の範囲】[Claims] 【請求項1】 係数データを用いて適応フィルタ処理を
行なう第一の信号処理プロセッサと、 前記係数データを格納し前記第一の信号処理プロセッサ
に出力する係数データ記憶手段と、 前記係数データを修正処理しこの修正処理した修正係数
データを前記係数データ記憶手段に格納する第二の信号
処理プロセッサとを備えることを特徴とする適応信号処
理装置。
1. A first signal processor for performing adaptive filter processing using coefficient data, coefficient data storage means for storing the coefficient data and outputting the coefficient data to the first signal processor, and modifying the coefficient data. An adaptive signal processing device, comprising: a second signal processor for processing and storing the corrected coefficient data that has undergone this correction processing in the coefficient data storage means.
JP13035092A 1992-05-22 1992-05-22 Adaptive signal processing unit Pending JPH06112854A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP13035092A JPH06112854A (en) 1992-05-22 1992-05-22 Adaptive signal processing unit

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP13035092A JPH06112854A (en) 1992-05-22 1992-05-22 Adaptive signal processing unit

Publications (1)

Publication Number Publication Date
JPH06112854A true JPH06112854A (en) 1994-04-22

Family

ID=15032295

Family Applications (1)

Application Number Title Priority Date Filing Date
JP13035092A Pending JPH06112854A (en) 1992-05-22 1992-05-22 Adaptive signal processing unit

Country Status (1)

Country Link
JP (1) JPH06112854A (en)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2017195561A (en) * 2016-04-22 2017-10-26 富士通株式会社 Digital signal processing circuit and signal processing apparatus including multiple digital signal processing circuits

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2017195561A (en) * 2016-04-22 2017-10-26 富士通株式会社 Digital signal processing circuit and signal processing apparatus including multiple digital signal processing circuits
US10090927B2 (en) 2016-04-22 2018-10-02 Fujitsu Limited Digital signal processing circuit and signal processing device that includes a plurality of digital signal processing circuits

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