JP2004172661A - Processing method and processing apparatus for audio signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To reduce a sound pressure at an undesirable position and in an undesirable direction when forming a sound field by a speaker array. <P>SOLUTION: A processing system is provided with: a plurality of digital filters DF0 to DFn to which audio signals are respectively supplied; and the speaker array 10. Supplying outputs of the digital filters DF0 to DFn to speakers SP0 to SPn of the speaker array 10 forms the sound field. Setting a prescribed delay time respectively to the digital filters DF0 to DFn forms points with a greater sound pressure than the sound pressure of surroundings and points with a smaller sound pressure than the sound pressure of surroundings to the sound filed. A low pass characteristic is provided to the frequency response of the digital filters DF0 to DFn. <P>COPYRIGHT: (C)2004,JPO

Description

[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to an audio signal processing method and apparatus suitable for use in a home theater or the like.
[0002]
[Prior art]
A speaker array suitable as a home theater or an AV system is a speaker array (for example, see Patent Document 1). FIG. 14 shows an example of the speaker array 10. The speaker array 10 includes a large number of speakers (speaker units) SP0 to SPn. In this case, as an example, n = 255 and the speaker diameter is several centimeters. Therefore, the speakers SP0 to SPn are actually arranged two-dimensionally on a plane. For simplicity, it is assumed that they are arranged on a horizontal straight line.
[0003]
Then, the audio signal is supplied from the source SC to the delay circuits DL0 to DLn and is delayed by a predetermined time τ0 to τn, and the delayed audio signal is supplied to the speakers SP0 to SPn through the power amplifiers PA0 to PAn, respectively. . The delay times τ0 to τn of the delay circuits DL0 to DLn will be described later.
[0004]
Then, in any place, the sound waves output from the speakers SP0 to SPn are synthesized, and the sound pressure resulting from the synthesis is obtained. Therefore, as shown in FIG. 14, in the sound field formed by the speakers SP0 to SPn, predetermined points Ptg and Pnc are set as follows.
Ptg: A place where the sound pressure is desired to be higher than the surroundings. Sound pressure boost point.
Pnc: A place where the sound pressure is desired to be lower than the surroundings. Sound pressure reduction point.
Then, the method of setting an arbitrary location as the sound pressure enhancement point Ptg can be roughly classified into the method shown in FIG. 15 or FIG.
[0005]
That is, in the case of the method shown in FIG.
L0 to Ln: distance from each speaker SP0 to SPn to sound pressure enhancement point Ptg
s: speed of sound
Then, the delay times τ0 to τn of the delay circuits DL0 to DLn are
τ0 = (Ln−L0) / s
τ1 = (Ln−L1) / s
τ2 = (Ln−L2) / s
...
τn = (Ln−Ln) / s = 0
Set to.
[0006]
Then, when the audio signal output from the source SC is converted into a sound wave by the speakers SP0 to SPn and output, the sound wave is output with a delay of time τ0 to τn shown in the above equation. Therefore, when those sound waves reach the sound pressure enhancement point Ptg, they all arrive at the same time, and the sound pressure of the sound pressure enhancement point Ptg becomes higher than the surroundings.
[0007]
That is, in the case of the system shown in FIG. 15, a time difference occurs between the sound waves due to a path difference from the speakers SP0 to SPn to the sound pressure enhancement point Ptg, and the time difference is compensated by the delay circuits DL0 to DLn to provide the sound pressure enhancement point Ptg. It focuses the sound. Hereinafter, this type of system is referred to as a “focus type”, and the sound pressure enhancement point Ptg is also referred to as a “focus”.
[0008]
In the case of the method shown in FIG. 16, the delay times τ0 to τn of the delay circuits DL0 to DLn are set so that the phase wavefronts of traveling waves (sound waves) output from the speakers SP0 to SPn are the same. Thus, the directivity is given to the sound wave, and the direction of the directivity is set to the direction of the sound pressure enhancement point Ptg. This system can also be considered as a case where the distances L0 to Ln are set to infinity in a focus type system. Hereinafter, this type of system will be referred to as a “directivity type”, and the direction of a sound wave at which the phase wavefronts of the sound waves are aligned will be referred to as a “directivity direction”.
[0009]
[Patent Document 1]
Japanese Patent Application Laid-Open No. 9-233591
[0010]
[Problems to be solved by the invention]
However, as described above, the main function of the speaker array 10 is to obtain the focus or directivity by the delay times τ0 to τn to realize the sound pressure enhancement point Ptg. At this time, the sound is supplied to the speakers SP0 to SPn. The amplitude of the audio signal only changes the sound pressure.
[0011]
Therefore, as a method of reducing the sound pressure at the sound pressure reduction point Pnc, it is conceivable to use the directivity of the speaker array 10. For example, a main pole (main lobe) is formed in the direction of the sound pressure enhancement point Ptg, and the sub pole (side lobe) is sufficiently reduced, or the directional characteristic is such that the direction of the sound pressure reduction point Pnc has null sensitivity. And so on.
[0012]
However, in order to do so, it is necessary to increase the number n of the speakers SP0 to SPn so that the size of the entire speaker array is sufficiently larger than the wavelength of the sound wave. However, this method is extremely difficult to realize in practice. Alternatively, the influence of the change in the sound pressure may reach the sound pressure enhancement point Ptg where the focus and the directivity are matched.
[0013]
Further, in a home theater or an AV system, it is necessary to consider multi-channel stereo. That is, with the spread of DVD players and the like, the number of multi-channel stereo sources is increasing. For this reason, it is necessary for the user to install speakers of the number of channels. However, this requires considerable space.
[0014]
The present invention is intended to solve the above problems.
[0015]
[Means for Solving the Problems]
In the present invention, for example,
Supply audio signals to multiple digital filters,
Each output of the plurality of digital filters is supplied to each of a plurality of speakers constituting a speaker array to form a sound field,
By setting a predetermined delay time for each of the plurality of digital filters, a first point having a larger sound pressure than the surroundings and a second point having a smaller sound pressure than the surroundings are formed in the sound field,
A method of processing an audio signal, wherein a low-pass filter characteristic is given to a frequency response of the audio signal at the second point by adjusting amplitude characteristics of the plurality of digital filters.
It is assumed that.
Therefore, a point at which the sound pressure is higher than the surroundings is set by setting the delay time of the digital filter, and a point at which the sound pressure is lower than the surroundings is set by the amplitude characteristics of the digital filter.
[0016]
BEST MODE FOR CARRYING OUT THE INVENTION
(1) Outline of the present invention
In the present invention, since the outputs of the speakers of the speaker array are combined in space and responded by respective pointers, the responses are interpreted as a digital filter in a pseudo manner. Then, a response signal at “a place Pnc where sound pressure is not desired to be heard as much as possible” is predicted, the amplitude is changed without changing the delay applied to each speaker, and the frequency characteristics are controlled in the manner of creating a digital filter.
[0017]
By controlling the frequency characteristic, the sound pressure at the place Pnc where the sound pressure is not desired to be heard is reduced as much as possible, and the band in which the sound pressure can be reduced is expanded. At this time, the sound pressure is reduced as naturally as possible.
[0018]
(2) Analysis of speaker array 10
Here, in order to simplify the description, a plurality of n speakers SP0 to SPn are arranged in a row in the horizontal direction to configure a speaker array 10, and the speaker array 10 is configured in a focus system shown in FIG. It is assumed that
[0019]
Here, it is considered that each of the delay circuits DL0 to DLn of the focus type system is realized by an FIR digital filter. Also, as shown in FIG. 1, it is assumed that the filter coefficients of the FIR digital filters DL0 to DLn are represented by CF0 to CFn, respectively.
[0020]
Then, it is considered that an impulse is input to the FIR digital filters DL0 to DLn and the output sound of the speaker array 10 is measured at points Ptg and Pnc. Note that this measurement is performed at the sampling frequency of the reproduction system including the digital filters DL0 to DLn or a higher sampling frequency.
[0021]
Then, the response signals measured at points Ptg and Pnc become a sum signal obtained by spatially propagating the sounds output from all speakers SP0 to SPn and acoustically adding them. At this time, for ease of explanation, it is assumed that signals output from speakers SP0 to SPn are impulse signals delayed by digital filters DL0 to DLn. In the following, the response signal added through this spatial propagation is referred to as “spatial combined impulse response”.
[0022]
Since the point Ptg sets the delay components of the digital filters DL0 to DLn for the purpose of making a focus here, the spatial combined impulse response Itg measured at the point Ptg is, as shown in FIG. It becomes a large impulse. Further, the frequency response (amplitude part) Ftg of the spatially synthesized impulse response Itg is flat in the entire frequency band as shown in FIG. Therefore, the point Ptg is a sound pressure enhancement point.
[0023]
In practice, the spatial combined impulse response Itg is accurate due to the frequency characteristics of the speakers SP0 to SPn, the frequency change during spatial propagation, the reflection characteristics of the wall in the middle of the road, and the time axis deviation specified by the sampling frequency. Although it is not an impulse, it is described here with an ideal model for simplicity.
[0024]
On the other hand, the spatially combined impulse response Inc measured at the point Pnc is considered to be a combination of impulses having time axis information, and is a signal in which the impulse is dispersed with a certain width as shown in FIG. You can see that. In FIG. 1, although the impulse response Inc at the point Pnc is a pulse train in which the pulse trains are arranged at equal intervals, generally, the interval between the pulse trains is random.
[0025]
At this time, information relating to the position of the point Pnc is not included in the filter coefficients CF0 to CFn, and the original filter coefficients CF0 to CFn are all based on the positive impulse. The frequency response Fnc of the impulse response Inc is also a composite of impulses in the positive direction.
[0026]
As a result, as is clear from the design principle of the FIR digital filter, as shown in FIG. 1, the frequency response Fnc is flat in a low frequency range and tends to attenuate at higher frequencies, that is, close to a low-pass filter. Will have properties. Also, at this time, the spatially synthesized impulse response Itg at the sound pressure enhancement point Ptg is one large impulse, but the spatially synthesized impulse response Inc at the point Pnc has a frequency response at the point Pnc because the impulse is dispersed. The level of Fnc is smaller than the level of frequency response Ftg at point Ptg. Therefore, the point Pnc is a sound pressure reduction point.
[0027]
Then, at this time, assuming that the spatially synthesized impulse response Inc is one spatial FIR digital filter, the FIR digital filter Inc originally has the amplitude value of the impulse including the time factor in the filter coefficients CF0 to CFn. Since the frequency response Fnc is changed by changing the contents (amplitude, phase, etc.) of the filter coefficients CF0 to CFn, the frequency response Fnc is changed. That is, the frequency response Fnc of the sound pressure at the sound pressure reduction point Pnc can be changed by changing the filter coefficients CF0 to CFn.
[0028]
From the above, when the delay circuits DL0 to DLn are configured by the FIR digital filters and their filter coefficients CF0 to CFn are selected, the sound pressure enhancement point Ptg and the sound pressure reduction point Pnc can be set in a place where a sound field is required. Can be set to
[0029]
(3) Speaker array in a closed space
14 to 16, the sound field is an open space, but generally, as shown in FIG. 2, the sound field is a space or room RM acoustically closed by a wall WL or the like. In this room RM, the sound Atg output from the speaker array 10 is reflected on the wall surface WL around the listener LSNR by selecting the focal position Ptg or the directional direction of the speaker array 10, and then is transmitted to the listener LSNR. It can be focused.
[0030]
Then, in this case, even though the speaker array 10 is in front of the listener LSNR, sound can be heard from behind. However, in this case, since the sound Atg from the rear is the target sound, the sound Atg is set to be heard as loud as possible, and the sound Anc from the front is an unintended “leakage sound”. Must be set.
[0031]
For this purpose, as shown in FIG. 3, a virtual image of the entire room is considered based on the number of reflections of the sound Atg. Then, since this virtual image can be considered to be equivalent to the open space of FIG. 15 or FIG. 16, the position Ptg ′ of the virtual image corresponding to the sound pressure enhancement point Ptg is set at the position of the virtual image of the listener LSNR. The focus or directivity direction of the speaker array 10 is set. The sound pressure reduction point Pnc is set at the position of the actual listener LSNR.
[0032]
With the above configuration, it is possible to arrange virtual speakers behind and to the side of the multi-channel stereo without arranging speakers behind and to the side of the listener LSNR, thereby enabling surround stereo reproduction. Become.
[0033]
When the virtual speaker is realized by the focus type in this way, the position of the focus Ptg is set on the wall surface WL, not on the position of the listener LSNR, depending on the purpose, application, or the content of the source, or at any other place. Can also be set. In addition, the sense of localization of “where to hear from” cannot be strictly evaluated only by the sound pressure difference, but here it is considered important to increase the sound pressure.
[0034]
(4) Method of reducing sound pressure at point Pnc
In the room (closed space) RM shown in FIGS. 2 and 3, if the position of the listener LSNR is determined, the position of the sound pressure enhancement point Ptg is determined, and as a result, the delay time set by the filter coefficients CF0 to CFn is determined. When the position of the listener LSNR is determined, the position of the sound pressure reduction point Pnc is also determined. As shown in FIG. 4A, the position where the pulse of the spatial combined impulse response Inc at the sound pressure reduction point Pnc is determined is determined (FIG. 4A). Is the same as the spatially synthesized impulse response Inc of FIG. 1). Also, by changing the pulse amplitude values A0 to An in the digital filters DL0 to DLn, the controllable sample width (the number of pulses) becomes the sample width CN in FIG. 4A.
[0035]
Therefore, by changing the amplitudes A0 to An, the pulse (in the sample width CN) shown in FIG. 4A can be changed to a pulse (spatial composite impulse response) Inc ′ having a level distribution as shown in FIG. 4B, for example. As shown in FIG. 4C, the frequency response can be changed from the frequency response Fnc to the frequency response Fnc '.
[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 leakage sound Atn from the front is smaller than the target rear sound Atg, and the sound from the rear can be heard well.
[0037]
What is important at this time is that even if the amplitudes A0 to An are changed to form a pulse train such as the spatially synthesized impulse response Inc ', the spatially synthesized impulse response Itg and the frequency response Ftg of the sound pressure enhancement point Ptg change only in the amplitude value. That is, uniform frequency characteristics can be maintained. Therefore, according to the present invention, the frequency response Fnc 'is obtained at the sound pressure reduction point Pnc by changing the amplitudes A0 to An.
[0038]
(5) How to calculate spatially combined impulse response Inc '
Here, a method for obtaining a required spatially synthesized impulse response Inc ′ from the spatially synthesized impulse response Inc will be described.
[0039]
In general, when a low-pass filter is configured by an FIR digital filter, design methods using window functions such as Hamming, Hanning, Kaiser, and Blackman are well known, and the frequency response of a filter designed by these methods is relatively sharp. It is known that off characteristics can be obtained. However, in this case, since the pulse width that can be controlled by the amplitudes A0 to An is determined to be CN samples, design is performed using a window function within this range. Then, when the shape of the window function and the number of CN samples are determined, the cutoff frequency of the frequency response Fnc 'is determined.
[0040]
In this method, specific values of the amplitudes A0 to An are obtained from the window function and the CN sample. For example, as shown in FIG. By specifying the "coefficient", it is possible to specify the amplitudes A0 to An and perform the inverse calculation. In this case, a plurality of coefficients may affect one pulse in the spatially synthesized impulse response Inc, and if the number of corresponding coefficients (= the number of speakers SP0 to SPn) is small, FIG. As illustrated, there may be no corresponding coefficient.
[0041]
Preferably, the width of the window of the window function is substantially equal to the distribution width of the CN samples. When a plurality of coefficients affect one pulse in the spatially combined impulse response Inc, the coefficients may be distributed. Although this distribution method is not specified here, it is preferable to preferentially adjust the amplitude that has little effect on the spatially synthesized impulse response Itg and has a large effect on the spatially synthesized impulse response Inc ′.
[0042]
Further, as shown in FIG. 6, a plurality of points Pnc1 to Pncm can be set as the sound pressure reduction point Pnc, and the amplitudes A0 to An satisfying these points can be obtained by simultaneous equations. If the simultaneous equations are not satisfied, or if the amplitudes A0 to An that affect a specific pulse of the spatially synthesized impulse response Inc do not apply as shown in FIG. 5, the curve approximates to the target window function curve. Then, the amplitudes A0 to An can be obtained by the least square method or the like.
[0043]
Also, for example, the filter coefficients CF0 to CF2 correspond to the point Pnc1, the filter coefficients CF3 to CF5 correspond to the point Pnc2, the filter coefficients CF6 to CF8 correspond to the point Pnc3,. The relationship between the coefficients CF0 to CFn and the points Pnc1 to Pncm can be nested.
[0044]
Further, by devising the sampling frequency, the number of speaker units, and the spatial arrangement, it is possible to design such that the coefficient that influences each pulse of the spatially synthesized impulse response Inc is stochastically present as much as possible. It is. Further, as in the case of the discretization at the time of measurement, the spatially synthesized impulse response Inc passes through a space in which the sound radiated from the speakers SP0 to SPn is a continuous sequence, so strictly speaking, the coefficient that affects each pulse Is not specified as one, but here, for convenience, it is treated as such so as to be easily used as a guide during calculation. Experiments have confirmed that there is no practical problem even in this case.
[0045]
▲ 6 ▼ Example
(6) -1 First embodiment
FIG. 7 shows an example of a processing system according to the present invention, and FIG. 7 shows an audio signal line for one channel. That is, a digital audio signal is extracted from the source SC, the audio signal is supplied to the FIR digital filters DF0 to DFn through the variable high-pass filter 11, and the filter output is supplied to the speakers SP0 to SPn through the power amplifiers PA0 to PAn.
[0046]
In this case, since the cutoff frequency of the frequency response Fnc ′ can be estimated from the sample width CN of the controllable spatial synthesis impulse response Inc, the cutoff frequency of the variable high-pass filter 11 is linked to the cutoff frequency of the frequency response Fnc ′. Controlled. With this control, it is possible to pass the audio signal only in a band where the frequency response Ftg is superior to the frequency response Fnc ′. For example, in the case of FIG. 2, when the low-frequency portion of the frequency response Fnc 'is at the same level as the low-frequency portion of the frequency response Ftg, the effective band of the source is controlled, and the low-frequency portion is not used. Only those bands that are effective when heard can be output.
[0047]
The digital filters DF0 to DFn constitute the above-described delay circuits DL0 to DLn. Further, in the power amplifiers PA0 to PAn, the digital audio signal supplied thereto is D / A converted and then power-amplified or D-class amplified and supplied to the speakers SP0 to SPn.
[0048]
Then, in this case, for example, the routine 100 shown in FIG. 8 is executed in the control circuit 12, and the characteristics of the high-pass filter 11 and the digital filters DF0 to DFn are set as described above. That is, when the points Ptg and Pnc are input to the control circuit 12, the processing of the control circuit 12 starts from step 101 of the routine 100, and then in step 102, the delay times τ0 to τn in the digital filters DF0 to DFn are calculated. Subsequently, in step 103, the spatially combined impulse response Inc at the sound pressure reduction point Pnc is simulated, and the number of controllable samples CN is predicted.
[0049]
Then, in step 104, the cutoff frequency of the low-pass filter that can be created based on the window function is calculated. Next, in step 105, among the amplitudes A0 to An corresponding to each sample of the pulse train of the spatial synthesis impulse response Inc, The amplitudes A0 to An are obtained by listing up which amplitudes are valid. Then, in step 106, the cutoff frequency of the variable high-pass filter 11 and the delay times τ0 to τn of the digital filters DF0 to DFn are set according to the above results, and then the routine 100 is terminated in step 107.
[0050]
As described above, the sound pressure enhancement point Ptg and the sound pressure reduction point Pnc can be obtained.
[0051]
(6) -2 Second embodiment
In the system shown in FIG. 9, data of the cutoff frequency of the variable high-pass filter 11 and the delay times τ0 to τn of the digital filters DF0 to DFn are calculated for a plurality of points Ptg and Pnc. This is a case where the data is stored in the storage device 13 as a database.
[0052]
When the data of the points Ptg and Pnc are input to the storage device 12 during use of the reproducing system, the corresponding data is extracted from the storage device 13, and the cutoff frequency of the variable high-pass filter 11 and the delay of the digital filters DF0 to DFn. Times τ0 to τn are set.
[0053]
(6) -3 Third embodiment
In the system shown in FIG. 10, the digital audio signal from the source SC is processed by the variable high-pass filter 11 and the digital filters DF0 to DFn, for example, as described in (6) -1 (first embodiment). The processing result signal is supplied to the speakers SP0 to SPn through the digital addition circuit 14 and the power amplifiers PA0 to PAn.
[0054]
Further, the digital audio signal output from the source SC and the filter output of the variable high-pass filter 11 are supplied to a digital subtraction circuit 15 to extract a digital audio signal of a middle and low frequency component (a flat portion in FIG. 4C). It is. Then, the digital audio signal of the middle and low frequency components is supplied to the digital addition circuit 14 through the processing circuit 16.
[0055]
Therefore, the leak sound at the sound pressure reduction point Pnc can be controlled in accordance with the processing of the processing circuit 16.
[0056]
(6) -4 Fourth embodiment
FIG. 11 equivalently shows the processing contents of the FIR digital filters DF0 to DFn. A digital audio signal is supplied from the source SC to the original FIR digital filters DF0 to DFn through a fixed digital high-pass filter 17, and The filter output is supplied to the digital adding circuit 14. Further, a digital audio signal is supplied from the source SC to the processing circuit 16 through the digital low-pass filter 18.
[0057]
Therefore, when the processing of the processing circuit 16 can be realized by a digital filter, the processing can be executed by the digital filters DF0 to DFn.
[0058]
(6) -5 Fifth embodiment
FIGS. 12 and 13 show that one speaker array 10 realizes virtual speakers SPLF, SPRF, SPLB, and SPRB on the left front, right front, left rear, and right rear of the listener LSNR, thereby realizing a 4-channel surround stereo sound. This is where a field is formed.
[0059]
For this reason, as shown in FIG. 12, the speaker array 10 is arranged in front of the listener LSNR in the room RM. As shown in FIG. 13, for the left front channel, a digital audio signal DLF on the left front is extracted from the source SC, and this signal DLF is supplied to FIR digital filters DFLF0 to DFLFn through the variable high-pass filter 12LF. The filter output is supplied to speakers SP0 to SPn through digital addition circuits AD0 to ADn and power amplifiers PA0 to PAn.
[0060]
For the right front channel, a right front digital audio signal DRF is extracted from the source SC, and this signal DRF is supplied to FIR digital filters DFRF0 to DFRFn through a variable high-pass filter 12RF, and the output of the filter is supplied to a digital addition circuit. It is supplied to speakers SP0 to SPn through AD0 to ADn and power amplifiers PA0 to PAn.
[0061]
Further, the left rear channel and the right rear channel have the same configuration as the left front channel and the right front channel, and the description is omitted by changing the symbols LF and RF in the reference numerals to the symbols LB and RB.
[0062]
As described with reference to FIGS. 7 and 8, respective values are set for each channel. For the left front channel and the right front channel, for example, the virtual speakers SPLF and SPRF are realized by the system described with reference to FIG. For the left rear channel and the right rear channel, virtual speakers SPLB and SPRB are realized by, for example, the system described with reference to FIG. Therefore, these virtual speakers SPLF to SPRB form a 4-channel surround stereo sound field.
[0063]
According to the above-described system, surround multi-channel stereo can be realized by one speaker array 10, and a large amount of space is not required for installing speakers. Also, when increasing the number of channels, it is only necessary to add a digital filter, and there is no need to add speakers.
[0064]
▲ 7 ▼ Other
In the above description, a window function was used as a design guideline of the spatially synthesized impulse response Inc ′, and a relatively steep low-pass filter characteristic was formed. However, the desired characteristic was obtained by adjusting the amplitude of the coefficient using a function other than the window function. You may.
[0065]
In the above description, all the amplitudes of the filter coefficients are set to the positive-direction pulse trains, and the spatial synthesis impulse responses are all set to the positive-valued pulse trains. However, the delay for focusing on the sound pressure enhancement point Ptg is set. The characteristics of the sound pressure reduction point Pnc may be defined by setting the pulse amplitude in each filter coefficient in the positive or negative direction while maintaining the characteristics.
[0066]
Further, in the above description, the impulse is basically used as an element for adding a delay, but this is for the sake of simplicity, and this basic delay element is used as a tap of a plurality of samples having a specific frequency response. An effect can also be obtained. For example, a quasi-pulse sequence that can provide a pseudo-oversampling effect can be used as a basis. In this case, the coefficient has a negative component in the amplitude direction, but it can be said that the intended effect and execution means are the same.
[0067]
Further, in the above description, the delay with respect to the digital audio signal is represented by the coefficient of the digital filter. Furthermore, one or a plurality of combinations of the amplitudes A0 to An may be prepared, and the combination may be set as at least one of the sound pressure enhancement point Ptg and the sound pressure reduction point Pnc. When the speaker array 10 realizes a virtual rear speaker shown in FIG. 3, for example, and the application is fixed and a general reflection position or a listening position can be assumed, the filter coefficient is assumed in advance. Fixed filter coefficients CF0 to CFn corresponding to the sound pressure enhancement point Ptg and the sound pressure reduction point Pnc.
[0068]
Further, in the above description, when determining the amplitudes A0 to An of the filter coefficients corresponding to the spatially synthesized impulse response Inc ′, the influence of the attenuation by air is not considered. Simulation calculation can be performed by incorporating parameters. Further, it is also possible to measure each parameter by some measuring means, determine more appropriate amplitudes A0 to An, and perform a more accurate simulation.
[0069]
Further, in the above description, the speaker array 10 is a case where the speakers SP0 to SPn are arranged on a horizontal straight line, but they may be arranged on a plane or may be arranged with a depth. Further, it is not always necessary to arrange them in an orderly manner. In the above description, the focus type system is mainly described, but a similar process can be executed in the case of a directional type system.
[0070]
[List of abbreviations used in this specification]
AV: Audio and Visual
D / A: Digital to Analog
FIR: Finite Impulse Response
[0071]
【The invention's effect】
According to the present invention, when sound reproduction is performed by a speaker array, the sound pressure at a target place can be increased and the sound pressure at a specific place can be reduced. In this case, it is desired to reduce the sound pressure. Since a spatial window function is applied to the impulse response for the position and the direction and the synthesis is performed, it is possible to particularly reduce the response in the middle and high frequency range where the sense of arrival (localization sense) of the sound wave is easily perceived. At this time, there is no need to increase the scale of the required speaker array, and the practical students are high.
[0072]
Also, when configuring a multi-channel stereo, a surround multi-channel stereo can be realized by one speaker array, and a large amount of space is not required for installing speakers.
[Brief description of the drawings]
FIG. 1 is a diagram for explaining the present invention.
FIG. 2 is a diagram for explaining the present invention.
FIG. 3 is a diagram for explaining the present invention.
FIG. 4 is a diagram for explaining the present invention.
FIG. 5 is a diagram for explaining the present invention.
FIG. 6 is a diagram for explaining the present invention.
FIG. 7 is a system diagram illustrating one embodiment of the present invention.
FIG. 8 is a diagram for explaining the present invention.
FIG. 9 is a system diagram showing another embodiment of the present invention.
FIG. 10 is a system diagram showing another embodiment of the present invention.
FIG. 11 is a system diagram showing another embodiment of the present invention.
FIG. 12 is a diagram for explaining the present invention.
FIG. 13 is a system diagram showing another embodiment of the present invention.
FIG. 14 is a diagram for explaining the present invention.
FIG. 15 is a diagram for explaining the present invention.
FIG. 16 is a diagram for explaining the present invention.
[Explanation of symbols]
10: speaker array, DF0 to DFn: digital filter, DL0 to DLn: delay circuit, LSNR: listener, PA0 to PAn: power amplifier, Pnc: sound pressure reduction point, Ptg: sound pressure enhancement point, RM: room (closed space) ), SC: source, SP0 to SPn: speaker (speaker unit), WL: wall

Claims (8)

Supply audio signals to multiple digital filters,
The outputs of the plurality of digital filters are supplied to each of a plurality of speakers constituting a speaker array to form a sound field,
By setting a predetermined delay time for each of the plurality of digital filters, a first point having a larger sound pressure than the surroundings and a second point having a smaller sound pressure than the surroundings are formed in the sound field,
A method of processing an audio signal, wherein amplitude characteristics of the plurality of digital filters are adjusted to provide a low-pass filter characteristic to a frequency response of the audio signal at the second point. 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 reaches at least one of the first point and the second point. The method for processing an audio signal according to claim 1 or claim 2,
Processing of an audio signal in which when forming the first point and the second point in the sound field, filter coefficients of the plurality of digital filters are calculated and set for each of the plurality of digital filters; Method. The method for processing an audio signal according to claim 1 or claim 2,
Processing of an audio signal, wherein when forming the first point and the second point in the sound field, filter coefficients of the plurality of digital filters are retrieved from a database and set for each of the plurality of digital filters; Method. A plurality of digital filters to which audio signals are respectively supplied,
A speaker array configured by arranging a plurality of speakers,
Providing outputs of the plurality of digital filters to each of the plurality of speakers to form a sound field;
By setting a predetermined delay time for each of the plurality of digital filters, a first point having a larger sound pressure than the surroundings and a second point having a smaller sound pressure than the surroundings are formed in the sound field,
An audio signal processing device, wherein a low-pass filter characteristic is given to a frequency response of the audio signal at the second point by adjusting amplitude characteristics of the plurality of digital filters. The audio signal processing device according to claim 5,
An audio signal processing device configured to reflect sound waves output from the speaker array on a wall surface and then reach at least one of the first point and the second point. The audio signal processing device according to claim 5 or 6,
Processing of an audio signal in which when forming the first point and the second point in the sound field, filter coefficients of the plurality of digital filters are calculated and set for each of the plurality of digital filters; apparatus. The audio signal processing device according to claim 5 or 6,
Processing of an audio signal, wherein when forming the first point and the second point in the sound field, filter coefficients of the plurality of digital filters are retrieved from a database and set for each of the plurality of digital filters; apparatus.

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