JPH02237400A - Sound field correction device - Google Patents

Sound field correction device

Info

Publication number
JPH02237400A
JPH02237400A JP1058363A JP5836389A JPH02237400A JP H02237400 A JPH02237400 A JP H02237400A JP 1058363 A JP1058363 A JP 1058363A JP 5836389 A JP5836389 A JP 5836389A JP H02237400 A JPH02237400 A JP H02237400A
Authority
JP
Japan
Prior art keywords
filter
sound field
peak
field correction
correction filter
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.)
Granted
Application number
JP1058363A
Other languages
Japanese (ja)
Other versions
JPH07112320B2 (en
Inventor
Kazuhiro Nakamura
一啓 中村
Takeshi Miyagawa
猛 宮川
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.)
Panasonic Holdings Corp
Original Assignee
Matsushita Electric Industrial Co Ltd
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 Matsushita Electric Industrial Co Ltd filed Critical Matsushita Electric Industrial Co Ltd
Priority to JP1058363A priority Critical patent/JPH07112320B2/en
Publication of JPH02237400A publication Critical patent/JPH02237400A/en
Publication of JPH07112320B2 publication Critical patent/JPH07112320B2/en
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Abstract

PURPOSE:To obtain a natural sound in terms of listening sense by providing a peak correction filter to smooth only a steep peak in a transmission characteristic of a sound field correction filter. CONSTITUTION:An impulse response at a listening point measured in advance is inputted to an input terminal 1 of a filter coefficient calculation means 6 and a desired impulse response is inputted to an input terminal 2, then they are discrete Fourier transformed by an FFT(Fast Fourier Transform) unit 61. A filter characteristic calculator 62 receiving the output calculates the filter characteristic and the peak is corrected by a peak correction filter 63 and inverse discrete Fourier transformation is applied by an inverse FFT 64 to obtain a correction coefficient, which is set in the sound field correction filter 2 and the filter characteristic is obtained. Then the characteristic of a transfer function of the sound field correction filter 2 is preserved and only a steep peak is smoothed. Thus, a natural signal in terms of listening sense is reproduced from a speaker 5.

Description

【発明の詳細な説明】 産業上の利用分野 本発明は、車室内等の特定の場所の音場の伝達関数を、
所望の伝達関数に補正する音場補正装置に関する。
DETAILED DESCRIPTION OF THE INVENTION Field of Industrial Application The present invention provides a method for determining the transfer function of a sound field at a specific location such as a vehicle interior.
The present invention relates to a sound field correction device that corrects a desired transfer function.

従来の技術 第3図は従来の音場補正装置の構成を示すブロック図で
ある。第3図において、1はCDプレーヤ等の音響信号
発生器(ディジタル)であり、2の音場補正フィルタに
接続されている。2は、あらかじめ測定した音場の伝達
特性を補正するような特性をもったフィルタでる。3は
音場補正フィルタの出力をアナログに変換するD/A変
換器、4はアナログ信号3を増幅する増幅器、5は増幅
器4に接続されたスビーカである。
BACKGROUND OF THE INVENTION FIG. 3 is a block diagram showing the configuration of a conventional sound field correction device. In FIG. 3, 1 is an acoustic signal generator (digital) such as a CD player, which is connected to a sound field correction filter 2. In FIG. Reference numeral 2 denotes a filter having characteristics that correct the transfer characteristics of the sound field measured in advance. 3 is a D/A converter that converts the output of the sound field correction filter into an analog signal; 4 is an amplifier that amplifies the analog signal 3; and 5 is a speaker connected to the amplifier 4.

次に上記従来例の動作について説明する。第33ヘ−ジ 図において、音響信号発生器1から出力されたディジタ
ル信号は、音場補正フィルタ2に入力され、フィルタリ
ングされた後、D/A変換器3でアナログ信号に変換さ
れ、増幅器4にて増幅され、スピーカ5より放射される
Next, the operation of the above conventional example will be explained. In the 33rd Hage diagram, the digital signal output from the acoustic signal generator 1 is input to the sound field correction filter 2, and after being filtered, it is converted to an analog signal by the D/A converter 3, and then the digital signal is sent to the amplifier 4. The signal is amplified and radiated from the speaker 5.

ここで、スピーカ5のおかれた音場の、スピカ入力端子
から受聴位置までの伝達関数をH、音場補正フィルタの
入力から受聴位置まで所望の伝達関数をKとし,音場補
正フィルタ2への入力信号を■、音場補正フィルタ2の
伝達関数をA, D/A変換器33.−よび増幅器4の
伝達関数を1とすると、次式の関係が成シ立つ(ここで
H, K,  I,A等は周波数領域での表現)。
Here, the transfer function from the speaker input terminal to the listening position in the sound field in which the speaker 5 is placed is H, the desired transfer function from the input of the sound field correction filter to the listening position is K, and the transfer function is transferred to the sound field correction filter 2. , the input signal of the sound field correction filter 2 is A, and the D/A converter 33. - and the transfer function of the amplifier 4 is set to 1, then the following relationship holds true (here, H, K, I, A, etc. are expressed in the frequency domain).

H−A・I=K・工 ’    A=K/H このように、上記従来の音場補正装置でも、音場補正フ
ィルタの伝達関数Aを、上式のように設定することで音
場特性を補正することができる。
H-A・I=K・工' A=K/H In this way, even in the conventional sound field correction device described above, the sound field characteristics can be adjusted by setting the transfer function A of the sound field correction filter as shown in the above equation. can be corrected.

発明が解決しようとする課題 しかしながら、上記従来の音場補正装置では、補正対象
の伝遵関数Hが急激なディプを有する場合、そのディプ
を補正しようとするために、音場補正フィルタの伝達関
数Aは、特定の周波数にピークをもち、スピーカから再
生された信号は聴感上不自然になってしまうという問題
があった。
Problems to be Solved by the Invention However, in the conventional sound field correction device described above, when the transmission function H to be corrected has a sharp dip, in order to correct the dip, the transfer function of the sound field correction filter is A has a problem in that it has a peak at a specific frequency, and the signal reproduced from the speaker sounds unnatural.

本発明はこのような従来の問題を解決するものであり、
補正された信号が、聴感上不自然でないようにスピーカ
から再生できる優れた音場補正装置を提供することを目
的とするものである。
The present invention solves these conventional problems,
It is an object of the present invention to provide an excellent sound field correction device that can reproduce a corrected signal from a speaker so that it does not sound unnatural.

課題を解決するだめの手段 本発明は上記目的を達成するために、従来の音場補正装
置に、ピーク補正フィルタを設け、音場袖正フィルタの
伝達関数Aの特徴は保存し、急激なピークだけを平滑化
するようにしたものである。
Means for Solving the Problems In order to achieve the above object, the present invention provides a peak correction filter in a conventional sound field correction device, preserves the characteristics of the transfer function A of the sound field correction filter, and eliminates sudden peaks. It is designed to smooth only the

作    用 したがって、本発明によれば,音場補正フィルタの周波
数特性の急激なピークだけを補正することによって、そ
のピークを平滑化するここができ、また急激なピーク以
外は平滑化しないために、音場補正フィルタの周波数特
性を保存することがで5へ一ノ さ,聴感的により自然な信号をスビーカから再生するこ
とができるという効果を有する。
Therefore, according to the present invention, by correcting only the sharp peaks in the frequency characteristics of the sound field correction filter, it is possible to smooth the peaks, and since the peaks other than the sharp peaks are not smoothed, By preserving the frequency characteristics of the sound field correction filter, it is possible to reproduce an audibly more natural signal from the speaker.

実施例 第1図は、本発明の一実施例の構成を示すブロック図で
ある。第1図において、1は、CDプレヤ等のディジタ
ル音響信号発生器であり、その出力は音場補正フィルタ
2に接続されている。音場補正フィルタ2は、スピーカ
5から受聴点に至る音場の伝達関数を補正するためのデ
ィジタルフィルタで、FIR(Finite Impu
lse Response)フィルタで構成される。F
IRフィルタの係数は後述のフィルタ係数算出手段6で
計算し、設定する。
Embodiment FIG. 1 is a block diagram showing the configuration of an embodiment of the present invention. In FIG. 1, 1 is a digital acoustic signal generator such as a CD player, the output of which is connected to a sound field correction filter 2. In FIG. The sound field correction filter 2 is a digital filter for correcting the transfer function of the sound field from the speaker 5 to the listening point.
response) filter. F
The coefficients of the IR filter are calculated and set by a filter coefficient calculating means 6, which will be described later.

3は、音場補正された信号をアナログ信号に変換するD
/A変換器で,4は、そのアナログ信号を増幅する増幅
器、5は、増幅器4の出力を音に変換するスピーカであ
る。
3 is D for converting the sound field corrected signal into an analog signal.
In the /A converter, 4 is an amplifier that amplifies the analog signal, and 5 is a speaker that converts the output of the amplifier 4 into sound.

6は、フィルタ係数算出手段で、その入力端子■には、
スピーカの入力端子から受聴点迄のインパルス応答h(
1}が入力し、入力端子■には、音場6・\−・ 補正フィルタから受聴点に至る全系の所望のインするF
FT(Fast Fourier Transform
)で、それぞれh (ilおよびk (ilが入力し、
H(nlおよびK (n)に変換し出力する。62は、
フィルタ特性算出器で、次式の演算により、フィルタ特
性A (nlを出力する。
6 is a filter coefficient calculation means, and its input terminal ■ is
Impulse response h(
1} is input, and the input terminal ■ receives the desired input F of the entire system from the correction filter to the listening point.
FT (Fast Fourier Transform)
), respectively h (il and k (il input,
Convert to H(nl and K(n) and output. 62 is
The filter characteristic calculator outputs the filter characteristic A (nl) by calculating the following equation.

A (nl = K (nl / H (nlただし H (nl = X2 h (ii WNフ0 K (n) = ’X:2 k (i l WN斎;0 WN = e−”N 63は、ピーク補正フィルタで.A(nlの急激なピー
クだけを平滑化するために、ある周波数kにおけるA 
(nlの値をA (k+とし、次の演算を行う。
A (nl = K (nl / H (nlHowever, H (nl = X2 h (ii WNfu0 K (n) = ' A correction filter is used to smooth only the sharp peaks of .A(nl) at a certain frequency k.
(The value of nl is set to A (k+, and the following calculation is performed.

A(kl= ( bA(kl+c B )/d;(A(
kl≧aBのとき) = A (kl       ; ( A(kl< a
 Bのとき)7/\−7 ただし ここで、BはA (k+の前後におけるL個のA(1)
の平均値であり、a,  b,C,L  j+  L等
は音場の状態により適切な値を選定する。
A(kl=(bA(kl+cB)/d;(A(
When kl≧aB) = A (kl; (A(kl< a
B) 7/\-7 However, here, B is A (L A(1) before and after k+
, and appropriate values for a, b, C, L j+ L, etc. are selected depending on the state of the sound field.

上式は、A (k+がaBよシ大きい場合、A(k+を
それよシ小さな値に置き換えるものである064は、ピ
ーク補正されたA (k+を逆離散フーリ工変換する逆
FFTで、その出力は時間領域で表現された補正係数で
ある。この補正係数は音場補正フィルタ2の係数として
、あらかじめ設定される。
The above formula is, when A(k+ is larger than aB, A(k+ is replaced with a smaller value). The output is a correction coefficient expressed in the time domain. This correction coefficient is set in advance as a coefficient of the sound field correction filter 2.

次に、上記実施例の動作について説明する。Next, the operation of the above embodiment will be explained.

初めに、音場補正フィルタ2の係数を設定するときの動
作について説明する。
First, the operation when setting the coefficients of the sound field correction filter 2 will be explained.

フィルタ係数算出千段6の入力端子■に、あらかじめ測
定した受聴点のインパルス応答h(1}を入力し、入力
端子■に、所望のインパルス応答k(1)を入力すると
、これらはFFT61で離散フーリ工変換され,それぞ
れH (nl, K (n)となる。この出力を受けて
、フィルタ特性算出器62でフィルタ特性A (n.l
が計算され、ピーク補正フィルタ63でピーク補正され
た後、逆FFT64で逆離敬フリエ変換されて、補正係
数が求められる。この補正係数は音場補正フィルタ2の
中に設定され、フィルタ特性A(nlが定められる。
Input the impulse response h(1) of the listening point measured in advance to the input terminal ■ of the filter coefficient calculation stage 6, and input the desired impulse response k(1) to the input terminal ■.These are discretely processed by FFT61. They are Fourier transformed and become H (nl, K (n), respectively. Upon receiving this output, the filter characteristic calculator 62 calculates the filter characteristic A (n.l
is calculated, subjected to peak correction by a peak correction filter 63, and then subjected to inverse FFT 64 to obtain a correction coefficient. This correction coefficient is set in the sound field correction filter 2, and the filter characteristic A(nl) is determined.

次に、音場補正装置の動作について説明する。Next, the operation of the sound field correction device will be explained.

音響信号発生器1から出たティジタル信号I (nlは
、音場補正フィルタ2(伝達関数A (n.l )を通
り、D ./ A変換器3でアナログ信号に変換され,
増幅器4で増幅された後、スピーカ5から(伝達関数H
(nl)音響信号として受聴点に達する。
The digital signal I (nl) output from the acoustic signal generator 1 passes through the sound field correction filter 2 (transfer function A (n.l), and is converted into an analog signal by the D./A converter 3.
After being amplified by the amplifier 4, from the speaker 5 (transfer function H
(nl) Reaches the listening point as an acoustic signal.

従って受聴点での信号は、 H(nlA(nlI(nl=H(nlK(n)/H(n
l H I(n)= K (ni I (nl となり、所望の音場K (nl I (nlを得ること
ができる。
Therefore, the signal at the listening point is H(nlA(nlI(nl=H(nlK(n)/H(n
l H I(n)=K (ni I (nl), and the desired sound field K (nl I (nl) can be obtained.

このように上記実施例によれば、音場補正フィルタの周
波数特性の急激なピークが平滑化され、それ以、外の所
では平滑化されないため、聴感的に9ヘ−ン 目然な音声を再生出来るという効果を有する。
In this way, according to the above embodiment, the sharp peaks in the frequency characteristics of the sound field correction filter are smoothed, and other areas are not smoothed. It has the effect of being reproducible.

第2図は本発明の他の実施例の要部であるフィルタ係数
算出手段のブロック図である。この実施例では、ピーク
補正フィルタ63は、フィルタ特性A (n)のピーク
の補正ではなく,スピーカのおかれた音場の伝達特性H
(n)のディソプの補正、あるいは所望の伝達関数K 
(niのピークの補正を、第1の実施例と同様に行う。
FIG. 2 is a block diagram of filter coefficient calculating means, which is a main part of another embodiment of the present invention. In this embodiment, the peak correction filter 63 does not correct the peak of the filter characteristic A (n), but the transmission characteristic H of the sound field in which the speaker is placed.
Disop correction of (n) or desired transfer function K
(The peak of ni is corrected in the same way as in the first embodiment.

フィルタ特性A (n.lは、フィルタ特性算出千段6
2において、H(n)とK(nlより計算される。した
がって,この実施例でも、第1の実施例と同様の効果を
有する。
Filter characteristic A (n.l is filter characteristic calculation stage 6
2, it is calculated from H(n) and K(nl. Therefore, this embodiment also has the same effect as the first embodiment.

発明の効果 本発明は上記実施例よシ明らかなように,ピーク補正フ
ィルタを設けて音場補正フィルタの伝達特性における急
激なピークだけを平滑化するようにしたものであシ、し
たがって聴惑的によシ自然な音を聴取者に提供できると
いう効果を有する。
Effects of the Invention As is clear from the above embodiments, the present invention provides a peak correction filter to smooth only the sharp peaks in the transfer characteristics of the sound field correction filter. This has the effect of providing a more natural sound to the listener.

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

第1図は本発明の一実施例における音場補正装置のプロ
ノク図、第2図は本発明の他の実施例に10−\一/ おける同装置の要部ブロソクス、第3図は従来の音場補
正装置のブロック図である。 1・・・音響信号発生器、2・・・音場補正フィルタ、
3・・D/A変換器、4・・・宿幅器、5・・スピーカ
、6・・・フィルタ係数算出手段、61・・・FFT、
62・・・フィルタ特性算出器、63・・・ピーク補正
フィルタ、64・・・逆FFT0 代理人の氏名 弁理士 粟 野 重 孝 ほか1名一7
71−
FIG. 1 is a block diagram of a sound field correction device according to an embodiment of the present invention, FIG. 2 is a block diagram of the main part of the same device according to another embodiment of the present invention, and FIG. 3 is a diagram of a conventional sound field correction device. FIG. 2 is a block diagram of a sound field correction device. 1... Acoustic signal generator, 2... Sound field correction filter,
3...D/A converter, 4...Buffer device, 5...Speaker, 6...Filter coefficient calculation means, 61...FFT,
62...Filter characteristic calculator, 63...Peak correction filter, 64...Inverse FFT0 Name of agent Patent attorney Shigetaka Awano and 1 other person17
71-

Claims (2)

【特許請求の範囲】[Claims] (1)各インパルス応答信号をフーリエ変換する第一お
よび第二のFFTと、前記第一および第二のFFTの出
力の比を計算する計算手段と、前記計算手段の出力にお
ける急激なピークの値を平滑するピーク補正手段と、前
記ピーク補正手段の出力を逆フーリエ変換する逆FFT
とを備えた、フィルタ係数算出手段と、前記算出手段に
より算出された係数を設定したディジタルフィルタによ
り、ディジタル音源からの入力信号を補正して出力する
音場補正フィルタと、前記音場補正フィルタの出力をA
/D変換し、増幅してスピーカから音を出す手段とを備
えた音場補正装置。
(1) first and second FFTs that Fourier transform each impulse response signal; calculation means for calculating the ratio of the outputs of the first and second FFTs; and a value of a sudden peak in the output of the calculation means. and an inverse FFT that performs inverse Fourier transform on the output of the peak correction means.
a sound field correction filter that corrects and outputs an input signal from a digital sound source using a digital filter to which the coefficients calculated by the calculation means are set; output A
/D conversion, amplification, and outputting sound from a speaker.
(2)フィルタ係数算出手段は、各インパルス応答信号
をフーリエ変換する第一および第二のFFTと、前記第
一および第二の各FFT出力のピーク補正をする第一、
第二のピーク補正手段と、前記第一と第二のピーク補正
出力の比を計算する計算手段と、前記計算手段の出力を
逆フーリエ変換する逆FFTとから構成されることを特
徴とする請求項(1)記載の音場補正装置。
(2) The filter coefficient calculating means includes first and second FFTs that perform Fourier transform on each impulse response signal, and a first and second FFT that performs peak correction on each of the first and second FFT outputs.
A claim characterized in that it is comprised of a second peak correction means, a calculation means for calculating the ratio of the first and second peak correction outputs, and an inverse FFT that performs inverse Fourier transform on the output of the calculation means. The sound field correction device according to item (1).
JP1058363A 1989-03-10 1989-03-10 Sound field correction device Expired - Lifetime JPH07112320B2 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP1058363A JPH07112320B2 (en) 1989-03-10 1989-03-10 Sound field correction device

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP1058363A JPH07112320B2 (en) 1989-03-10 1989-03-10 Sound field correction device

Publications (2)

Publication Number Publication Date
JPH02237400A true JPH02237400A (en) 1990-09-19
JPH07112320B2 JPH07112320B2 (en) 1995-11-29

Family

ID=13082234

Family Applications (1)

Application Number Title Priority Date Filing Date
JP1058363A Expired - Lifetime JPH07112320B2 (en) 1989-03-10 1989-03-10 Sound field correction device

Country Status (1)

Country Link
JP (1) JPH07112320B2 (en)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH06178397A (en) * 1992-11-30 1994-06-24 Victor Co Of Japan Ltd Method for controlling sound image localization
US5404406A (en) * 1992-11-30 1995-04-04 Victor Company Of Japan, Ltd. Method for controlling localization of sound image

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH06178397A (en) * 1992-11-30 1994-06-24 Victor Co Of Japan Ltd Method for controlling sound image localization
US5404406A (en) * 1992-11-30 1995-04-04 Victor Company Of Japan, Ltd. Method for controlling localization of sound image

Also Published As

Publication number Publication date
JPH07112320B2 (en) 1995-11-29

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