JPH04132499A - Sound image controller - Google Patents

Abstract

PURPOSE:To control a sense of direction of a sound image independently of a listening position by controlling a phase characteristic of plural phase shifters so that a synthesized wave front of a reproduced sound of all speakers is independent of its frequency but depending on an angle inputted to a wave front angle input means. CONSTITUTION:A phase characteristic controller controls the phase characteristic of plural phase shifters 104-106 based on a wave front angle inputted to a wave front angle input means 101. Then plural speakers 107-109 arranged at a predetermined position reproduce outputs of the phase shifters 104-106. In this case, the phase characteristic controller controls the phase characteristic so that the synthesized wave front of the reproduced sound from all the speakers 107-109 is independent of the frequency but depending on an angle inputted to the wave front angle input means 101. thus, a listener receives an impression as if sound were radiated from a direction of wave front and the sense of direction is controlled.

Description

DETAILED DESCRIPTION OF THE INVENTION Field of the Invention The present invention relates to a sound image control device that controls a dynamically changing sound image.

BACKGROUND OF THE INVENTION In recent years, technological trends in the audio visual field have been changing from conventional stereo playback to systems that dynamically control sound images in accordance with images.

As a conventional technology, U.S. Patent No. 374679
There is a Dolby surround active matrix type sound image control device shown in No. 2, No. 3832886, and No. 3959590.

A conventional sound image control device will be described below with reference to the drawings.

First, I will explain how to encode Dolby Surround.

FIG. 5 is a block diagram showing the configuration of a Dolby Surround encoder.

In FIG. 5, 501 is an L (left channel) signal input terminal, 502 is an R (right channel) signal input terminal, and 50
3 is the C (center channel) signal input terminal, 504 is the S (
surround channel) signal input terminal, 505 is an attenuator that attenuates the C signal by 3 [dB], 506 is an attenuator 50
507 is an attenuator 50 which adds the output of 5 to the L signal.
An adder 508 adds the output of 5 to the R signal, and 508 adds the output of 5 to the R signal.
[dB] attenuator; 509 is a band-pass filter that passes the output of the attenuator 508 from 100 [Hz] to 7 [kHz]; 510 is a modified B-type noise reduction encoder that encodes the output of the band-pass filter 509; 511 is a phase shifter that creates a signal having a +90[deg] phase difference with respect to the output of the modified B-type noise reduction encoder 510; 512 is a +90[deg] phase difference of the phase shifter 511;
an adder 51 that adds the eg output to the output of the adder 50B;
3 is the -90[deg] output of the phase shifter 511 and the adder 50
7, an adder 514 outputs the output of the adder 512 as an Lt (encoder left channel) signal, and 515 outputs the output of the adder 513 as an Rt (encoder left channel) signal.
This is the Rt signal output terminal that outputs the encoder right channel) signal.

The operation of the Dolby Surround encoder configured as above will be explained.

The left channel) signal that is input to the Dolby surround encoder is sent to the speaker placed in front of the left of the listening position in the listening room, the R (right channel) signal is sent to the speaker placed in the front right, and the C (center channel) signal is sent to the speaker placed in front of the right. The signal is a mixed signal on the premise that the signal is reproduced by a speaker placed in the front, and the surround channel signal S is reproduced by two speakers placed at the left and right in the rear. C
The signal is attenuated by 3 [dB] by the attenuator 505 and then sent to the adder 5.
The signal is added to the L signal in step 06, and added to the R signal in adder 507, respectively. The S signal is attenuated by 3 [dB] by an attenuator 508,
Furthermore, a bandpass filter 509 filters 100 [Hz
[Bandwidth limited to kHz. Bandpass filter 509
The output of the modified B-type noise reduction encoder 510
encoded in . This encoding will be discussed later. The output of the modified B-type noise reduction encoder 510 is shifted by +90[deg] by a phase shifter 511, and added to the output of the adder 506 by an adder 512. Adder 5
The output of No. 12 becomes the Dolby surround encoder output Lt. Similarly, the output of the modified B-type noise reduction encoder 510 is phase-shifted by -90[deg] by the phase shifter 511, and added to the output of the adder 507 by the adder 513 to become the Dolby surround encoder output Rt.

The above processing can be summarized as equations (1) and (2).

Lt=L+o, 7C+0.7jS −(1)R
t=R+0.7C-0,7j S -(2) Here, j represents (-1)1'2, and the phase rotation is 90[
deg].

The modified B-type noise reduction encoder 510 has an amplitude frequency characteristic that changes depending on the level of the input signal. By decoding this encoded signal, it is possible to reduce the high frequency components of noise generated in the transmission media. Table 1 shows the amplitude frequency characteristics of the modified B-type noise reduction encoder using the input signal level as a parameter.

Next, the decoder of the Dolby Surround Active Matrix type sound image control apparatus will be explained in Table 1. Table 1 (Continued) FIG. 6 is a block diagram showing the configuration of the decoder of the Dolby Surround Active Matrix type sound image control apparatus.

In FIG. 6, 601 is an input terminal for the encoder output t (left channel) signal, and 602 is the encoder output Rt.
(Right channel) signal input terminal, 603 is Lt and Rt
an input balance control device for adjusting the balance between the
605 is a level control device that adjusts the absolute level of the balance-adjusted signals Lt and Rt, and 605 is the signal Lt that has been adjusted in absolute level.

Rt (left channel) signal, R (right channel)
606 is a delay device that delays the S signal output from the adaptive matrix 605; 607 is the delayed S signal 7[k];
Hz] low-pass filter that passes signals below 60
8 is a modified B-type noise reduction decoder that attenuates the noise of components below 7 [kHz] of the S signal; 609 is an L signal output from the adaptive matrix 605;

610 is a listening room; 611 is a mask level control device that controls the levels of the R signal, the C signal, and the S signal output from the modified B-type noise reduction decoder 608;
is placed to the right of the listening position in the listening room.
A speaker 612 that reproduces the R signal output from the mask level control device 809 is placed in front of the listening position in the listening room, and a speaker 613 that reproduces the C signal output from the mask level control device 609 is placed in front of the listening position in the listening room. A speaker 61 disposed at the front left of the listening position and reproducing the signal output by the mask level control device 609;
A speaker 4 is placed at the right rear of the listening position in the listening room and reproduces the S signal outputted by the mask level control device 609. A speaker 615 is placed at the left rear of the listening position in the listening room and is a mask level control device 609. This is a speaker that reproduces the S signal output by the.

The operation of the Dolby surround active matrix sound image control device decoder configured as described above will be described.

Equation (1) is input to the Dolby Surround Active Matrix sound image control device decoder.

These are encoder outputs Lt and Rt expressed by equation (2).

The input balance control device 603 receives the input signal Lt.

Adjust the balance of Rt. A level control device 6.4 adjusts the absolute levels of input signals Lt and Rt.

In the adaptive matrix 605, the input signal Lt.

The four output signals R, C, and 5 (7) are controlled according to the level difference of Rt. Therefore, the above-mentioned input signal Lt
It is necessary to adjust the balance and absolute levels of , Rt. The processing of the adaptive matrix 605 will be described in detail later. The delay device 6o6 is an adaptive matrix 6
05 S (surround channel) signal from 15 to 30[
msl delay.

The low-pass filter 607 filters the delayed S signal at 7 [kHz
Passes signals below 1. The modified B-type noise reduction decoder 608 reduces high frequency noise generated in the transmission medium included in the S signal. Table 2 shows the amplitude frequency characteristics using the input level of the modified B-type noise reduction decoder 6o8 as a parameter. The first characteristic of the decoder is
The characteristics are the opposite of those of the encoder shown in the table.

Table 2 Table 2 (continued) The mask level control device 609 outputs the output signals of the adaptive matrix 605 (left channel) signal, R (right channel) signal, C (center channel) signal, and a modified B-type noise reduction decoder. S output by 608
(Surround channel) This is a 4-channel volume that controls the signal level.

R signal output by mask level control device 609.

The C signal, L signal, and S signal are reproduced by speakers 611 to 615 arranged in the listening room.

Here, the adaptive matrix 605 will be explained.

FIG. 7 is a block diagram showing the configuration of the adaptive matrix 605. In FIG. 7, 701 is an Lt input terminal, 702 is an Rt input terminal, 703 is a band pass filter that limits the signal bands of Lt and Rt, and 704 is L'
(band-limited Lt) and R' (band-limited R
705 is L'.
a subtracter that takes the difference between
6 to 709 are full-wave rectifier circuits that perform full-wave rectification of L', R', C', and S', 710 is a logarithmic difference circuit that outputs the logarithmic difference DLR between R'' and L', and 711 is S 712 is a threshold switch that determines whether the output of the logarithmic difference circuit 710 or 711 is within a predetermined range; 713 is a determination by the threshold switch 712; Depending on the result, the time constant is 22 [ms] or 448 [rn
The output D of the logarithmic difference circuit 710 is a low-pass filter of
A dual time constant circuit 714 that processes LR has a time constant of 22 [msl] according to the determination result of the threshold switch 712.
Or a dual-time constant circuit that processes the output Dcs of the logarithmic difference circuit 711 with a low-pass filter of 448 [ms], 715
716 is a polarity dividing circuit that outputs EL and ER obtained by multiplying the output of the bi-temporal constant circuit 713 by a coefficient corresponding to its polarity, and 716 is a polarity dividing circuit that outputs Ec+Es obtained by multiplying the output of the bi-temporal constant circuit 714 by a coefficient corresponding to its polarity. A polarity division circuit 717 outputs the outputs EL and ER of the polarity division circuit 715 and the output Ec of the polarity division circuit 718.

ELLIEL by controlling the input signals Lt and Rt by Es.
RI ERLI Ellllll ECLI E
e*+ ESLI Voltage controlled amplifier that outputs ESR, 718 is the output of voltage controlled amplifier 717 E LL r
E LRrE RL rE RRJ CL・E CR+
ESLI Est+ and input signals Lt, Rt are multiplied by a predetermined constant and added, and L, R, C,
It is a connection network that outputs S.

The operation of the adaptive matrix configured as described above will be explained below.

The adaptive matrix 605 calculates the difference in the logarithms of the signal levels for the LR axis or the C8 axis, and detects from which direction the signal is dominant based on this difference. and,
By outputting signals in the dominant direction as they are and attenuating signals in other directions, the sense of direction of the reproduced sound is emphasized.

The bandpass filter 703 converts the input signals Lt and Rt to 100
[Hzl~7 [Limit the band to kHzl.

The outputs L' and R' of the bandpass filter 703 are expressed by equation (1),
As shown in equation (2), the main components are the encoder input signal and the R signal, respectively. Additionally, adder 704.
The output of the subtractor 705 is expressed by equation (3).

It is expressed by equation (4).

C”=C+0.7 (L + R) ... (3
)s' =-j S+0. 7 (L-R)
(4) From equations (3) and (4), it can be seen that c' and s' each have a C2S signal as their main component.

Each of L', R', C', and S' is full-wave rectified by full-wave rectifier circuits 706-709. After full-wave rectification, the pairs of L', RL and c', s' are processed by logarithmic difference circuits 710 and 711, respectively, and outputted as 5. .

Dcs is obtained. The processing of the logarithmic difference circuits 710 and 711 is expressed by equations (5) and (6), respectively.

DLR= 1 o ga (R'/L')
・(5) Dcs= l o JL (S'/C'
) ・(6) DLR is LR with respect to LR axis
Dcs indicates which of C8 is dominant with respect to the C8 axis.

The threshold switch 712 outputs the adaptive matrix 605 when the level difference between L and R1 or C and S is large, and short time constants 22 of dual time constant circuits 713 and 714 are used to quickly change R, C, and S. [ms
], and conversely, when the level difference is small, a long time constant of 484 [ms] is selected, and R, C, and S are changed gradually.

Threshold switch 712 is connected to logarithmic difference circuit 710 .
If the outputs DLRI and Dcs of 711 are both within the range of threshold level Lth, the bitemporal constant circuit 71
A time constant of 484 [ms] of 3,714 is selected, and if either one is outside the range, a time constant of 22 [ms] is selected. Table 3 shows the time constant ([m5l) selected by the Threnjord switch 712.

Table 3 In order to realize Table 3, the Threnjord switch 71
2 is the output D LRI of the logarithmic difference circuits 710 and 711
It consists of four comparators that compare Dcs and Threshold Level Lth, and an AND circuit that takes the logical sum of the comparator outputs.

The dual time constant circuit 713 integrates the output DLR of the logarithmic difference circuit 710 with a time constant of 484 or 22 [ms] depending on the determination result of the Threnjord switch 712. The integration circuit requires an RC integrator or one with transient characteristics equivalent to it. The bitemporal constant circuit 714 is also a logarithmic difference circuit 71
Similar processing is performed for the output Dcs of 1.

The polarity dividing circuit 715 is integrated by the bi-time constant circuit 713. It generates control voltages EL and E for the subsequent voltage control amplifier 717 according to the polarity of the voltage control amplifier 717. control voltage EL,
ER is expressed by equations (7) and (8).

EL=DLRDLR<O o DLR>O-(7) ER: ODLR<0 DLRDLR>O...18) Similarly, the control voltage E for the voltage control amplifier 717 generated by the polarity dividing circuit 716. +Es is the formula (9).

It is expressed by equation (10).

Ec" Dcs Dcs<0 0 Dcs> O-(9) Es"ODcs<0-DcsDcs>O-(10) The voltage control amplifier 717 outputs the output EL of the polarity dividing circuit 715.
, ER and the output Ec of the polarity dividing circuit 716.

By controlling the input signals Lt and Rt by Es, E LLI
Eub ERL+ EIIJII EcL+
Outputs EcR+EsL+E5R. here,
The input signal controlled by the output E of the polarity divider circuit is E
Let us express it as XY. FIG. 8 shows the relationship between the control voltage and amplification factor of the voltage controlled amplifier.

Coupling network 718 connects the output of voltage controlled amplifier 717 E LL+ ELTo ERL+ ERR+
ECL+ Ecp+ESLI ERR and input signal L
t and Rt in the ratio shown in Table 4, L, R,
Output C and S.

Table 4 Table 4 (continued) Table 4 (continued) In the output stage of the adaptive matrix 605, 25 [d
B] or more separation between channels can be ensured.

As described above, the Dolby surround active matrix type sound image control device decoder detects which direction of the signal is dominant with respect to the LR axis or the C8 axis.

When a dominant direction is detected, the signal in that direction remains unchanged, and the signals in directions other than the deviation are attenuated, thereby emphasizing the sense of direction of the reproduced sound. Therefore, sounds with a clear direction, such as dialogue, provide a clear sense of direction. On the other hand, if no dominant direction is detected, a normal stereo effect is obtained.

Problems to be Solved by the Invention However, in the conventional configuration described above, the direction of the sound image is determined by the relationship between the listening position and the speaker position.

Therefore, there is a problem in that when the listening position changes, the sound image direction also changes accordingly.

The present invention solves the above-mentioned conventional problems, and aims to provide a sound image control device that can control the sense of direction regardless of the listening position.

Means for Solving the Problems In order to solve the above problems, the sound image control device of the present invention includes a wavefront angle input means for inputting a wavefront angle, and a phase control device for controlling the phase characteristics of a plurality of phase shift devices based on the wavefront angle. a characteristic control device; a plurality of phase shift devices capable of changing phase characteristics in accordance with the phase characteristic control device and processing input signals; and a plurality of speakers that reproduce the output, and the phase characteristic control device controls a plurality of phase shifts so that the composite wavefront of the sound reproduced by all the speakers becomes the angle input to the wavefront angle input means regardless of the frequency. Controls the phase characteristics of the device.

According to the above-described configuration, the synthesized wavefront of sounds reproduced by all the speakers becomes the angle input to the wavefront angle input means, regardless of the frequency, under the control of the phase characteristic control device. Then, the listener receives the impression that the sound is being radiated from the direction in which the wave front arrives, and can control the sense of direction. Furthermore, by continuously changing the wavefront angle, the sense of movement of the sound image can be controlled.

EXAMPLE Hereinafter, an example of the present invention will be described with reference to the drawings.

FIG. 1 is a block diagram showing the configuration of a sound image control device in an embodiment of the present invention. In FIG. 1, 101 is a wavefront angle input means, and 102 is a phase shift device 1 according to the wavefront angle.
a phase characteristic control device for controlling phase characteristics of 04 to 106;
103 is a signal input terminal, 104 to 106 are phase shifters that change the phase of the Kugata signal in accordance with the control of the phase characteristic control device 102, and 107 to 109 are arranged at predetermined positions, each of which is a phase shifter. This is a speaker that reproduces the output signals 104 to 106. In this embodiment, for the sake of simplicity, it is assumed that all the speakers are arranged on the same plane in a grid pattern with regular intervals d [m].

The operation of the sound image control device of this embodiment configured as described above will be described below.

First, wavefront angle information is input to the wavefront angle input means 101 . Here, it is assumed that the angle between the wavefront and the plane on which the speaker is arranged is input as the wavefront angle information. The phase characteristic control device 1゜2 operates a phase shift device based on the angle between the wave front and the plane on which the speakers are arranged, and the coordinates of each speaker (in this embodiment, the coordinate unit is [m]). 104 to 106 are calculated.

FIG. 2 shows the plane on which the speakers are placed and the wavefront that is synthesized. Assume that the speakers are arranged on a grid on the plane y=O (xz plane) at intervals of d [ml]. At this time, FIG. 2 shows the state of the wavefront synthesized on the plane Z=O (Xy plane).

Let the angle between the wavefront and the plane y:o be θ[radl. The wave front and the direction of travel of the wave front are orthogonal. Therefore, adjacent speakers have a phase difference φ shown in equation (11).

φ=2・π*(dosinθ)/λ・(N)λ=
c/f...(I
2) Here, λ is the wavelength ([mco), and C is the speed of sound ([m/
s ]), f is the frequency of the sound ([Hz]). Since each adjacent speaker has a phase difference given by equation (II), from the leftmost speaker, i:o, L ・
..., if we assign a number as n, the phase difference φ1 that the i-th speaker should have with the 0-th speaker as the reference is (+3
) is given by the formula.

φ+=2・i 4・(cls i nθ)/λ −(
+3) The phase frequency characteristic of the phase shift device corresponding to the i-th speaker is expressed by equation (13).

Further, it is desirable that the amplitude frequency characteristics of all phase shift devices be flat. Therefore, the transfer function H: (f) of the phase shift device corresponding to the i-th speaker is expressed as (14) % Equation % When Equation (14) is inversely Fourier transformed, the impulse response h1(t) of Equation (+5) becomes can get.

h:(t)=(π4a+t)l/hria+t)]
= i + 5), a = (iIId@Sinθ) / c - (l
(15) is illustrated in FIG. As shown in Figure 3, equation (15) is expressed as ±1/(
It is attenuated using π·(a+t)1 as an envelope. Therefore, equation (15) may be approximated by truncating it at an appropriate length with t=-a as the center.

From the above, the phase characteristic control device 102 calculates the characteristic given by equation (15) for the first phase shift device based on the input wavefront angle θ and the speaker interval d. If the phase shift device is an FIR filter, the phase characteristic control device 102 will output the impulse response given by equation (15) as a filter coefficient.

The phase shift devices 104 to 106 may be any filter such that equation (15) becomes an impulse response. Specifically, it can be realized by an FIR filter that receives a signal from the signal input terminal 103 and uses equation (15) as a coefficient.

The speakers 107 to 109 each have a phase shifter 104
Play the output of ~106.

FIGS. 4(a) and 4(b) show examples of equal phase surfaces of sound fields synthesized by the sound image control device of the present invention. Figure 4 (a
), (b), 21 omnidirectional speakers were arranged on the X-axis at an interval of d=0.18 [m] with x=O as the center. The wavefront angle was set to π/6[radl (=30[deg]), and a solid line was drawn at the location where the phase at a certain moment was O~π[radl, and the location where π~2π[radl was left blank.

The signal frequency f in FIG. 4(a) is 500 [Hzl], and the signal frequency f in FIG. 4(b) is 1 [kHz2]. The equiphase front can be considered to represent a wavefront synthesized by the sound image control device.

As can be seen from Figures 4(a) and (b), in the range x = -1,8-+1.8[m] where the speakers are located, regardless of the frequency, the wavefront is It forms an angle of π/6 [radl with respect to the X axis. That is, it can be seen that the wavefront direction is controlled.

In a sound field with a wavefront at a certain angle, a time difference occurs between the ears of a listener who listens to the wavefront. Furthermore, the frequency characteristics of the binaural sound pressure also differ due to head diffraction.

Humans synthesize this information to obtain a sense of direction of sound.
One gets the impression that a sound source exists in the direction of arrival of the wave front.

FIGS. 4(a) and 4(b) show wavefronts in a plane parallel to the xy plane. When the speakers are arranged on a grid rod, the sound emitted from the speakers having the same X coordinate value forms a cylindrical wavefront with a wavefront parallel to the Z axis.

Therefore, the wavefront of any plane parallel to the X-y plane is
The wavefronts are the same as those shown in FIGS. 4(a) and 4(b).

Furthermore, assume that the wavefront angle input to the wavefront angle input means 101 is changed, for example, as θ(1) in equation (+7).

θ(t)=π/2−a@jan (a/l) river (17)
Here, a is a constant. At this time, θ(1) is the wavefront angle of the sound emitted by the sounding body moving at a constant speed on the X-axis. By performing wavefront control using this θ(1), the listener feels that the sound image is moving at a constant speed on the X-axis.

Note that this example shows an example of synthesizing a wavefront perpendicular to the It can also be synthesized. In this case, the sense of direction in the vertical direction can be controlled. Furthermore, it is also possible to perform phase manipulation in any line segment direction on the NX-2 plane.

Effects of the Invention As described above, according to the present invention, the synthesized wavefront of sounds reproduced by all the speakers becomes the angle input to the wavefront angle input means, regardless of the frequency, by the control of the phase characteristic control device. Therefore, the listener receives the impression that the sound is being radiated from the direction in which the wave front arrives, and the effect is that the sense of direction can be controlled. Furthermore, since the same sense of direction is obtained within the space that is the interior of the rectangular parallelepiped whose bottom surface is the part where the speakers are arranged, the service area is wide. Furthermore, by continuously changing the wavefront angle, it is also possible to control the sense of movement of the sound image.

[Brief explanation of drawings]

FIG. 1 is a block diagram showing the configuration of a sound image control device according to an embodiment of the present invention, and FIG. 2 is a plan view showing a plane in which speakers are arranged and a wavefront to be synthesized in the same embodiment.
Figure 3 is a waveform diagram showing an example of the impulse response of a phase shifter, Figures 4 (a) and (b) are plan views showing an example of a combined wavefront, and Figure 5 shows the configuration of a Dolby Surround encoder. 6 is a block diagram showing the configuration of a decoder of a Dolby surround active matrix sound image control device, FIG. 7 is a block diagram showing the internal configuration of the adaptive matrix in FIG. 6, and FIG. 8 is a voltage control diagram. FIG. 3 is a characteristic diagram showing the relationship between control voltage and amplification factor of an amplifier. 101...Wavefront angle input means, 102...Phase characteristic control device, 103...Signal input terminal, 104
~106-... Phase shift device, 107-109...
speaker. Name of agent: Patent attorney Akira Okaji and two others, Isu I (R Speaker Figure 2 1, -d, -1 Notes $4 1! 薯・) Seventy crabs v@-

Claims (2)

[Claims] (1) A wavefront angle input means for inputting a wavefront angle; a phase characteristic control device for controlling the phase characteristics of a plurality of phase shifters according to the wavefront angle; and a phase characteristic of an input signal according to the phase characteristic control device. and a plurality of speakers arranged at predetermined positions to reproduce the output of the plurality of phase shifters, respectively, and the phase characteristic control device reproduces the output of all the speakers. A sound image control device that controls phase characteristics of a plurality of phase shift devices so that a synthesized wavefront of sound has an angle input to the wavefront angle input means regardless of frequency. (2) The sound image control device according to claim 1, wherein the wavefront angle is continuously changed.