JP2004172661A5

JP2004172661A5 -

Info

Publication number: JP2004172661A5
Application number: JP2002332565A
Authority: JP
Filing date: 2002-11-15
Publication date: 2005-03-17
Anticipated expiration: 2022-11-15

Description

【００２３】
なお、実際には、各スピーカＳＰ0〜ＳＰnの周波数特性、空間伝播時の周波数特性変化、行路途中の壁の反射特性、サンプリング周波数によって規定される時間軸のずれなどにより、空間合成インパルス応答Ｉtgは正確なインパルスとはならないが、ここでは簡単のため、理想的なモデルで記している。[0023]
In practice, the space synthesis impulse response Itg is determined by the frequency characteristics of each of the speakers SP0 to SPn, the change in frequency characteristics during space propagation, the reflection characteristics of the wall along the way, the deviation of the time axis defined by the sampling frequency, etc. It is not an accurate impulse, but for simplicity this is an ideal model.

【００３６】
つまり、図４Ｃおいて斜線を付けた部分の帯域分だけ音圧低減点Ｐncにおける音圧が低下することになる。したがって、図２の場合には、目的とする後方音Ａtgに対し、前方からの漏れ音Ａncが少なくなり、後方からの音が良く聞こえることになる。[0036]
That is, the sound pressure at the sound pressure reduction point Pnc is reduced by the band of the hatched portion in FIG. 4C. Therefore, in the case of FIG. 2, the leak sound Anc from the front decreases with respect to the target rear sound Atg, and the sound from the rear can be heard well.

【００６８】
さらに、上述においては、空間合成インパルス応答Ｉnc'に対応するフィルタ係数の振幅Ａ0〜Ａnを決定するとき、音波の伝搬時の空気による減衰の影響や、反射物による位相変化などのパラメータを盛り込んで、シミュレーション計算することもできる。また、何らかの測定手段により、それぞれのパラメータを測定して、より適切な振幅Ａ0〜Ａnを決定し、より正確なシミュレーションを行うこともできる。[0068]
Furthermore, in the above description, when determining the amplitudes A0 to An of the filter coefficients corresponding to the spatial synthetic impulse response Inc ', parameters such as the influence of air attenuation during sound wave propagation and the phase change due to reflectors are included. , Simulation can also be calculated. Also, it is possible to measure each parameter by some measurement means to determine more appropriate amplitudes A0 to An and to perform more accurate simulation.

Claims

The audio signal is supplied to each of a plurality of digital filters,
The outputs of the plurality of digital filters are supplied to each of the plurality of speakers constituting the speaker array to form a sound field,
A predetermined delay time is set in each of the plurality of digital filters so that the propagation delay times of the audio signal reaching the first point in the sound field via the respective digital filters and the respective speakers coincide with each other. And
A method of processing an audio signal, comprising adjusting the amplitude characteristics of the plurality of digital filters so as to give a low pass filter characteristic to the combined response of the audio signal at a second point in the sound field.

In the audio signal processing method according to claim 1,
A method of processing an audio signal, wherein a sound wave output from the speaker array is reflected on a wall surface and then reached at least one of the first point or the second point.

In the method of processing an audio signal according to claim 1 or 2,
When forming the first point and the second point in the sound field, processing of an audio signal in which filter coefficients of the plurality of digital filters are obtained by calculation and set in each of the plurality of digital filters Method.

In the method of processing an audio signal according to claim 1 or 2,
When forming the first point and the second point in the sound field, processing an audio signal in which filter coefficients of the plurality of digital filters are extracted from a database and set in each of the plurality of digital filters Method.

With multiple digital filters, each supplied with an audio signal,
The outputs of the plurality of digital filters are supplied to each of the plurality of speakers constituting the speaker array to form a sound field,
A predetermined delay time is set in each of the plurality of digital filters so that the propagation delay times of the audio signal reaching the first point in the sound field via the respective digital filters and the respective speakers coincide with each other. And
An audio signal processing apparatus that adjusts the amplitude characteristics of the plurality of digital filters so as to give low-pass filter characteristics to the combined response of the audio signal at the second point in the sound field.

In the apparatus for processing an audio signal according to claim 5,
An apparatus for processing an audio signal according to claim 1, wherein a sound wave output from the speaker array is reflected by a wall surface and then reaches at least one of the first point and the second point.

In the audio signal processing device according to claim 5 or 6,
When forming the first point and the second point in the sound field, processing of an audio signal in which filter coefficients of the plurality of digital filters are obtained by calculation and set in each of the plurality of digital filters apparatus.

In the audio signal processing device according to claim 5 or 6,
When forming the first point and the second point in the sound field, processing an audio signal in which filter coefficients of the plurality of digital filters are extracted from a database and set in each of the plurality of digital filters apparatus.