JP3233228B2 - Sound image localization device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound image localization apparatus capable of correctly localizing a sound image of a musical tone at an arbitrary position with a small number (about two) of speakers.

[0002]

2. Description of the Related Art When a human hears an audio signal such as a musical sound, the human can recognize the position of the sound source aurally. This is based on the difference between the characteristics of the signals reaching the left and right ears. In other words, when the sound source is located at a position shifted left and right from the front of the listener, a difference occurs in the time to reach the left and right ears. Also, there is a difference in the acoustic characteristics (impulse response) between the case where the sound source is in front of the listener and the case where the sound source is behind. This is due to the influence of the skull, earlobe, and the like.

[0003]

However, conventional electronic musical instruments and the like do not have a function for realizing the above-described characteristics for localizing the sound image of the sound source, and merely control the left and right volume levels. Met. For this reason, sound image localization was unnatural.

[0004] It is an object of the present invention to provide a sound image localization apparatus capable of clearly positioning a sound image with a simple configuration.

[0005]

SUMMARY OF THE INVENTION The present invention includes an output means for outputting a stereo signal through the input musical tone signal to the left and right of the filter, in order to obtain an impulse response waveform from a plurality of directions of the point, A table in which filter coefficients corresponding to a plurality of points are stored corresponding to each of the left and right filters , and a table separately for each of the left and right filters
To obtain the points of the impulse response waveform, versus the left and right filter based on each direction of the set <br/> left
A set means for setting each of the left and right filter the filter coefficients are read from the table to respond, and further comprising a.

[0006]

In the sound image localization apparatus of the present invention, the sound image is localized by reproducing the impulse response at the sound image localization position. Usually reads an impulse response waveform from the same localization position for the left and right channels (ears) (in
Although localizing the Pick) sound image filter coefficients for implementing a pulse response, in the present invention to read out the impulse response waveform from separate sound image localization position at the left and right channels (point).
This makes it possible to realize a unique sense of localization that has never been seen before.

[0007] The present invention can also be configured as follows. That is, a plurality of delay circuits that receive a tone signal of the same sound source and delay each by a predetermined time,
A sound image localization device including a plurality of FIR filters to which a plurality of signals extracted from the respective delay circuits are input, wherein the delay circuit determines a blank time until a direct sound of the impulse response of the sound source arrives. The FIR filter has a characteristic of realizing a response characteristic of the impulse response of the sound source until the arrived direct sound is attenuated. In this case, the delay time is realized by the delay circuit until the direct sound arrives.
The number of stages of the FIR filter can be greatly reduced, and the configuration can be simplified.

[0008]

FIG. 1 is an overall block diagram of a sound image localization apparatus according to an embodiment of the present invention. This is a device that outputs an input audio signal as a binaural signal so as to be localized at a predetermined position. The operation of this device is controlled by the CPU 10. The CPU 10 has a ROM 12 and an R
AM13, panel switch 14, display 15, MIDI
Interface 16 and interface 17 are connected.

The ROM 12 stores a control program and a transfer function to be given to an FIR filter described later.
The panel switch 14 includes a switch group for inputting a localization position (left / right channel, distance, direction) of the sound image. A MIDI controller (electronic musical instrument) such as a keyboard is connected to the MIDI interface 16. When the MIDI controller is used as a man-machine interface, a control signal is input via the MIDI interface 16. The pan circuit 21 is connected to the interface 17. Pan circuit 21
Is a circuit for applying delay, reverberation, and the like to the input audio signal to process it into a binaural signal localized at a predetermined position. The A / D converter 20 is connected to the pan circuit 21. The A / D converter 20 is a circuit that converts a plurality of streams of audio signals input from an external input terminal into digital signals. In the pan circuit 21, the tone signal is divided into binaural signals (L and R two-channel signals).
The signals of each channel are respectively supplied to D / A converters 22L and 22L.
The signal is converted into an analog signal by the 2R and input to the sound systems 23L and 23R. Sound system 23
L and 23R amplify this signal and output it from a speaker (headphone).

Here, as shown in FIG. 2, assuming that the sound source S is at the point P and the listener's both ears are at the positions of eL and eR.
The impulse response from the sound source S to both ears (eL, eR) is
For example, they are as shown in FIGS. In this way, no signal time (time until a sound directly reaches the ear) τL, τR is generated according to the distance from the impulse generation to the ear. In the pan circuit 21, this non-signal time is realized by the delay circuit 50.

The sections a-b and cd from the arrival of the direct sound to the attenuation of the indirect sound are represented by FIR filters 51,5.
2 is used to represent the response waveform. In the pan circuit 21, as shown in FIG. 4, the section of τL, τR is represented by one system of delay circuit 50, and the section of ab, cd is represented by FIR filters 51, 52 of left and right separate systems. I have.

FIG. 4 is a diagram showing a basic configuration of the pan circuit 21. The delay circuit 50 can be composed of, for example, a multi-stage shift register provided with a read tap for each stage. The CPU reads input signals from taps corresponding to the delay times τl and τr of the left channel and the right channel, and inputs the read signals to the FIR filters 51 and 52. The number n of tap stages is obtained by nL = (τL / c) sf nR = (τR / c) sf (c: sound speed, sf: sampling frequency).

FIG. 5 is a view showing a filter coefficient table stored in the ROM 12. The filter coefficients are
F representing an impulse response for localizing a sound image at a certain point
This is a transfer function of the IR filter. The table is provided for each of the left and right channels (HR, HL). The coefficients use different coordinate systems (x-yR, x-yL) with the ear as the origin for each of the left and right channels (ears). In both the left and right channels, every short distance D1 and long distance D2, 30 ° from the front (P0 = P′0 = 0 °, P1 = P′1)
= 30 °, P2 = P′2 = 60 °,..., P11 = P′1
24 coefficients are stored for each of the 12 angles (1 = 330 °). Here, Pn and P'n are both arguments representing angles, but one is marked with (') since the coordinate system (origin) differs between the left and right ears. One for each of the left and right channels is read out and set in the FIR filter based on the operation of the panel switch 14 and the control signal input from the MIDI interface 16. A unique effect can be obtained by reading the transfer function for localizing the sound image at different positions for each of the left and right channels.

FIG. 6 is a diagram showing a configuration of the pan circuit 21. As shown in FIG. The pan circuit 21 is obtained by adding a volume control circuit and a reverb circuit to the basic configuration shown in FIG.
The pan circuit 21 includes a plurality of similar circuits (modules) connected in parallel. One or a plurality of tone signals distributed by the selector 30 are input to each module. When a plurality of tone signals are input to one module, the plurality of signals are added and output in the selector 30.

The module 31 is provided with two FIR filters 32, 33 and a reverb circuit 40 externally, and a delay circuit 34, filters 35, 36, 3
7. Built-in multipliers M1 to M5 and adders A1 to A5. Delay circuit 34 (corresponding to delay circuit 50 in FIG. 4)
A) is composed of a delay line having two or more variable taps. The signal input to the delay circuit 34 is output as OL and OR after a predetermined time delay. This O
L and OR are FIR filters 32 and 33 (the FIR filters in FIG. 4).
(Corresponding to the filters 51 and 52) . The delay time of the delay circuit 34 is determined by the CPU 10 based on the control data. A predetermined transfer function is set in the FIR filters 32 and 33. The signals MIR and MIL that have been subjected to the convolution operation by the transfer function are input to filters 36 and 37. The filters 36 and 37 are composed of first- and second-order low-pass filters, and have a function of determining the frequency characteristics of the left and right direct sounds due to air or a shield. M
Reference numerals 2 and M3 denote multipliers for controlling the gains of the filters 36 and 37. The multiplier is input from the CPU 10.

Here, the tone signal is directly input from the selector 30 to the filter 35. This filter 35 → multiplier M1 → adder A1 → reverb circuit 40 → multiplier M4
The system from M5 to the adders A2 and A3 is a system for adding an indirect sound to a tone signal. That is, it is a circuit that expresses sound that is diffusely reflected in the space and remains as reverberation. For this reason, the filter 35 is configured by a first- or second-order low-pass filter that filters a low-frequency portion in which attenuation due to diffuse reflection is small. M1 is a multiplier for controlling the gain of the filter 35. The multiplier is input from the CPU 10. A1 is an adder that adds the F0 output (indirect sound output before reverb is added) other than the module (module A) to the indirect sound of module A and outputs the result to the reverb circuit 40.
In other words, the indirect sound is a circuit for expressing the sound because the indirect sound becomes a mixed sound due to reduced separability of a plurality of sound sources due to diffuse reflection or the like. Reverb circuit 40 provides reverberation time due to diffuse reflection. M4 and M5 are multipliers for controlling the amount of indirect sound given to the left and right channels. The multiplier is C
Provided from PU10.

This indirect sound is added to adders A2 and A3.
It is added to the musical sound of the direct sound system (system that has passed through the FIR filters 32 and 33). After that, adders A4 and A5
At the tone signal of module B (OUTL, OUT
R) are added, and D /
Output to the A converters 22L and 22R. The OUTL and OUTR of the module B are connected to the OUTL and OUTR of the module C.
Since L and OUTR are added, and OUTL and OUTR of the lower modules are sequentially added, OUTL and OUTR of module A are obtained by combining the outputs of all modules.

Selector signal, number of L and R stages of delay,
The filter coefficients of various filters and the coefficients of each multiplier are
As described above, the signal is supplied from the CPU 10 via the interface.
Two R filters and a reverb circuit are provided.
The IR filter and the reverb circuit may be operated in a time-sharing manner so as to be shared by a plurality of modules. Further, the outputs of the modules may not be sequentially added but all outputs may be added at one time.

FIGS. 7 and 8 are flowcharts showing the operation of the sound image localization apparatus. FIG. 7 shows the main routine. When the power of the apparatus is turned on, first, an initialization operation is executed (n1). The initialization operation includes resetting a register, reading preset data, and the like. Next, the panel processing (n2) and other processing (n3) are repeatedly executed.

FIG. 8 is a flowchart showing the panel processing operation. First, the mode is determined (n10). In mode 1 (normal mode: localizing a sound image at one point on a plane), the setting of the distance and direction of the position where the sound source (sound image) is to be localized is received (n11). The input distance and direction are those to the center of the face (the origin O in FIG. 2). The distance and direction (rL, θ) from the sound source to the left and right ears (eL, eR in FIG. 2) based on the input distance and direction.
L, rR, θR) are calculated (n12). On the other hand, in the mode 2 (the sound image is localized at different positions on both the left and right channels), the distance and direction from the position where the sound source is localized on the left channel to the left ear and the sound source are localized on the right channel. The input of the distance and direction from the position to the right ear is received (n13). Thereafter, the flow advances to n14.

At n14, the number of OL and OR stages and the multipliers M2 and M3 of the delay circuit 34 are calculated from the set distance. The multipliers M2 and M3 are calculated based on the distance rL (rR). rL ≦ D0 (midpoint of D0 is D1, D2: see FIG. 2) in the case of calculated in HL1 (θL) × (D1 / rL), rL> D0 HL 2 (θL) × in the case of (D2 / R
L) Calculate in ² . Further, the coefficients of the left and right filters are read out based on the distance and direction (n15). The various data obtained by the above operations are sent to the pan circuit 21 (n16). Various data are input to the designated module and the FIR filter and reverb circuit of the module. Thereafter, other panel processing (n17) is executed and the routine returns. Other panel processing includes setting operations such as module selection, direction / distance setting, data calculation, transmission, and selector control (which series is output to which module).

As described above, in this embodiment, since two types of transfer functions are provided according to the distance from the sound source to the ear,
A sense of distance in acoustic characteristics can also be realized. Further, the number of stages may be increased not only by two types of short distance / long distance but also by further subdivision. Also, by allowing the sound image to be localized at different positions in the left and right channels, it is possible to produce a unique effect not found in natural musical instruments.

In this embodiment, both the left and right signals are passed through the delay circuit 34 in accordance with the distance to the sound source. You may make it delay.

In the pan circuit shown in FIG. 6, filters for directly controlling the sound characteristics are separately provided for both the left and right channels (36, 37), but before the separation into the left and right channels (before the delay 34), one filter is provided. It may be provided in a system.

A crosstalk cancel circuit may be provided between the pan circuit and the D / A converter. Further, in this embodiment, two-dimensional localization has been described, but the present invention can be applied to three-dimensional localization.

[0026]

As described above, according to the present invention, an impulse response from different points (sound image localization positions) on the left and right channels is reproduced, so that a unique sound image localization feeling that has not been achieved in the past can be realized.

[Brief description of the drawings]

FIG. 1 is a block diagram of a sound image localization apparatus according to an embodiment of the present invention.

FIG. 2 is a view for explaining a localization function of the sound image localization apparatus.

FIG. 3 is a diagram illustrating a localization function of the sound image localization apparatus.

FIG. 4 is a diagram illustrating a localization function of the sound image localization apparatus.

FIG. 5 is a view showing an FIR filter coefficient table stored in the sound image localization apparatus.

FIG. 6 is a configuration diagram of a pan circuit of the sound image localization apparatus.

FIG. 7 is a flowchart showing the operation of the sound image localization apparatus.

FIG. 8 is a flowchart showing the operation of the sound image localization apparatus.

[Explanation of symbols]

21-pan circuit, 34-delay circuit, 32, 33-F
IR filter.

Claims (1)

(57) [Claims] 1. A and output means for outputting the entered tone signal as a stereo signal through left and right filter, to obtain an impulse response waveform from a plurality of directions of the point
A table storing filter coefficients corresponding to a plurality of points corresponding to each of the left and right filters; and an impulse at different points in each of the left and right filters.
To obtain a scan response waveform, each of right and left set
The filter coefficients corresponding to the left and right filters are read from the table based on the direction, and
A sound image localization device, comprising: setting means for setting each .