JP2017076949A - Method of generating reverberation sound in consideration of air absorption of sound - Google Patents
Method of generating reverberation sound in consideration of air absorption of sound Download PDFInfo
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- JP2017076949A JP2017076949A JP2015213432A JP2015213432A JP2017076949A JP 2017076949 A JP2017076949 A JP 2017076949A JP 2015213432 A JP2015213432 A JP 2015213432A JP 2015213432 A JP2015213432 A JP 2015213432A JP 2017076949 A JP2017076949 A JP 2017076949A
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本発明は、ステレオ信号の左の音声信号(L)と右の音声信号(R)の差の成分から12dB/octのローパスフィルタを使って残響音の信号を近似的に生成する方法に関するものである。 The present invention relates to a method of approximately generating a reverberant signal from a difference component between a left audio signal (L) and a right audio signal (R) of a stereo signal using a 12 dB / oct low-pass filter. is there.
多くのサラウンド方式があるが、音の空気吸収については考慮されていない。 There are many surround systems, but sound air absorption is not considered.
ステレオ信号の左の音声信号(L)と右の音声信号(R)の差の成分L−R(R−L)は通常反射音と考えられている。同時に、この差成分L−R(R−L)には直接音が多く含まれている。しかし、音は周波数の高い成分ほど、また距離が大きいほど空気によって吸収されるので、反射音は演奏会場が大きいほど高い周波数成分が失われている。それで、L−R(R−L)の高域成分を演奏会場の大きさに合わせて適度に減衰させてやる必要がある。 A component LR (RL) of the difference between the left audio signal (L) and the right audio signal (R) of the stereo signal is considered to be a normal reflected sound. At the same time, the difference component L-R (R-L) contains many direct sounds. However, since the sound is absorbed by the air as the component has a higher frequency and the distance increases, the higher the frequency component is lost in the reflected sound, the larger the performance venue. Therefore, it is necessary to moderately attenuate the high frequency component of LR (RL) according to the size of the performance hall.
音は周波数の2乗に比例して空気吸収されると一般に考えられている。それで、周波数の2乗に比例して減衰する12dB/octのローパスフィルタで高域を減衰させてやるなら、いっそう残響音に近くなると考えられる。また、ローパスフィルタの遮断周波数を可変にすることにより、演奏会場の大きさに合わせた高域の減衰を実現できる。図1は、ステレオ信号の左の音声信号(L)と右の音声信号(R)の差の成分L−R(R−L)を12dB/octのローパスフィルタで高域を減衰して残響音の信号を近似的に生成する本発明を表したものである。 It is generally considered that sound is absorbed by air in proportion to the square of the frequency. Therefore, if the high band is attenuated with a 12 dB / oct low-pass filter that attenuates in proportion to the square of the frequency, it is considered that the sound is more like reverberation. In addition, by making the cut-off frequency of the low-pass filter variable, it is possible to realize high-frequency attenuation that matches the size of the performance venue. FIG. 1 shows a reverberation sound obtained by attenuating a high frequency range of a difference component LR (RL) between a left audio signal (L) and a right audio signal (R) of a stereo signal with a 12 dB / oct low-pass filter. 2 represents the present invention for approximately generating the following signal.
試聴を繰り返した結果、残響時間T(秒)の演奏会場の残響音の信号の生成に最適な遮断周波数は、その周波数の音が空気吸収で3dB減衰するのに要する時間TcがTc≒0.25T(秒)となるようにして求めることができる。この時、遮断周波数と同じ周波数の音が空気吸収で3dB減衰するのに要する距離は音速を340(m/秒)とすると340Tc(m)となるが、図2のような音の空気吸収のデータ(例えば、JIS Z 8378)から逆算して遮断周波数を決定することができる。図3は、残響時間T(秒)が1.2(秒)、2.0(秒)、3.6(秒)の場合に、この方法で遮断周波数を求めたものである。 As a result of repeated trial listening, the optimum cut-off frequency for generating a reverberation sound signal at a performance venue with a reverberation time T (seconds) is a time Tc required for the sound of that frequency to be attenuated by 3 dB by air absorption, Tc≈0. It can be determined to be 25T (seconds). At this time, the distance required for the sound having the same frequency as the cut-off frequency to be attenuated by 3 dB by air absorption is 340 Tc (m) when the sound speed is 340 (m / sec). The cut-off frequency can be determined by calculating back from the data (for example, JIS Z 8378). FIG. 3 shows the cutoff frequency obtained by this method when the reverberation time T (seconds) is 1.2 (seconds), 2.0 (seconds), and 3.6 (seconds).
本発明は次のような効果を奏する。
(1)ステレオ信号のLとRの差成分L−R(R−L)を、12dB/octのローパスフィルタを使って高域を減衰させてやることにより、いっそう残響音に近づけることができる。The present invention has the following effects.
(1) The difference component LR (RL) between the L and R of the stereo signal can be made closer to reverberant sound by attenuating the high range using a 12 dB / oct low-pass filter.
(2)Tc≒0.25T(秒)の時間は、音場の成長過程で定常状態の約97%に達するのに要する時間であり、近似的に分散音場が生成されると考えられる。(音場のエネルギー密度の成長の割合は、時間t、残響時間Tを用いて、1−exp(−13.8t/T)と表せるが、t=0.25Tを代入して0.97を得ることができる。)(2) The time of Tc≈0.25 T (seconds) is the time required to reach about 97% of the steady state in the sound field growth process, and it is considered that a dispersed sound field is generated approximately. (The growth rate of the energy density of the sound field can be expressed as 1-exp (−13.8 t / T) using time t and reverberation time T, but 0.97 is obtained by substituting t = 0.25T. Can be obtained.)
(3)本発明を用いたシステムによって生成される残響音とメインスピーカーLとRの音の和によって演奏会場の音場が再現できる。(3) The sound field of the performance venue can be reproduced by the sum of the reverberant sound generated by the system using the present invention and the sounds of the main speakers L and R.
以下、本発明の実施の形態を図に基づいて説明する。 Hereinafter, embodiments of the present invention will be described with reference to the drawings.
図4は、本発明を簡単に実現するための一例で、トランスを用いてステレオ信号のLとRから差の成分L−R(R−L)を生成する回路と、コイルとコンデンサを使った12dB/octのローパスフィルタを組み合わせて、ステレオ信号のLとRから近似的に残響音の信号を生成する回路である。トランスには一次インピーダンス1200Ω、二次インピーダンス8Ωの出力トランス(SANSUIST−32)を用い、その性能を引き出すためにヘッドフォンアンプの出力を用いている。この実施例では、L(mH)とC(μF)の値は3段階に切り替えることができる。図5には、この実施例で用いたL(mH)とC(μF)の値の3段階の値とその時の遮断周波数が示されているが、それぞれ大中小の演奏会場の大きさ(残響時間T)に対応している。また、ローパスフィルタの抵抗値は設計を簡単にするためにクオリティファクタQ≒0.7(バターワースフィルタ)になるように設定したが、図6が示しているように空気吸収の特性を十分に近似している。図7は、この実施例で用いたスピーカーの構成であるが、L−R(R−L)のスピーカーは、リア側又はフロント側だけを鳴らしても効果がある。 FIG. 4 is an example for easily realizing the present invention, which uses a circuit that generates a difference component LR (RL) from L and R of a stereo signal using a transformer, a coil, and a capacitor. This circuit generates a reverberant signal approximately from L and R of a stereo signal by combining a low-pass filter of 12 dB / oct. An output transformer (SANSUIST-32) having a primary impedance of 1200Ω and a secondary impedance of 8Ω is used as the transformer, and the output of the headphone amplifier is used in order to extract the performance. In this embodiment, the values of L (mH) and C (μF) can be switched between three levels. FIG. 5 shows three levels of L (mH) and C (μF) values used in this example and the cut-off frequencies at that time. Corresponds to time T). In addition, the resistance value of the low-pass filter is set to be a quality factor Q≈0.7 (Butterworth filter) for simplicity of design, but the air absorption characteristics are sufficiently approximated as shown in FIG. doing. FIG. 7 shows the configuration of the speaker used in this embodiment, but the LR (R−L) speaker is effective even when only the rear side or the front side is sounded.
L ステレオ信号の左の音声信号。
R ステレオ信号の右の音声信号。
T(秒) 演奏会場の残響時間。
Tc(秒) 遮断周波数と同じ周波数の音が空気吸収で3dB減衰するのに要する時間。
L(mH) ローパスフィルタのコイルのインダクタンス。
C(μF) ローパスフィルタのコンデンサのキャパシタンス。L Audio signal to the left of the stereo signal.
R Audio signal to the right of the stereo signal.
T (seconds) Reverberation time at the performance venue.
Tc (seconds) Time required for a sound having the same frequency as the cutoff frequency to be attenuated by 3 dB due to air absorption.
L (mH) The inductance of the low-pass filter coil.
C (μF) Capacitance of the low-pass filter capacitor.
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CN110753297B (en) * | 2019-09-27 | 2021-06-11 | 广州励丰文化科技股份有限公司 | Mixing processing method and processing device for audio signals |
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