JPH06178398A - Method for controlling sound image localization - Google Patents

Abstract

PURPOSE:To make the calculation of a transmission characteristic of a signal converting circuit excelelnt with a small circuit scale and to improve sound quality by obtaining the transmission characteristic of the signal converting circuit based on a measured transfer function and taking average of width corresponding to critical band width. CONSTITUTION:A pair of microphones ML and MR are arranged at the both ears of a dummy head DM, measurement sound from a speaker SP is received and lease sound refL, refR and sound to be measured L, R are synchronized and recorded in a recorder DAT. A prescribed processing is executed on a workstation through the use of the recording and the frequency response IR(S) of a measurement position HRTF is obtained. Then, average moving processing is performed by optimum band width in accordance with a frequency band and it is defined as the transmission characteristic of the signal converting circuit. The moving averaged discrete frequency response is reverse FET- converted and defined as the time response of a cancel filter. Then, a moving average is executed so that the convergence of time response is quickened, the transfer characteristic of the signal converting circuit is made well calculate with a small scale, a cost is reduced and sound quality is improved.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention reproduces a signal processed by a plurality of signal conversion circuits to which the same sound source is supplied from a plurality of transducers arranged apart from each other, and an actual transducer (speaker). The present invention relates to a sound image localization control that makes a user feel that a sound image is localized at a desired arbitrary position different from the above, and particularly to an improvement in calculation of data for sound image localization control (transfer characteristics of a signal conversion circuit).

[0002]

2. Description of the Related Art Conventionally, there is a sound image localization method using a binaural technique in which a sound source is felt at a specific position (direction) by the level difference and phase difference (time difference) of signals in both ears. As a sound image localization method using an analog circuit, for example, Japanese Patent Application Laid-Open No. 53-140001 (Japanese Patent Publication No.
58-3638) There is a sound image localization method described in the official gazette. This is because the analog filter emphasizes / attenuates the level of a specific frequency band (amplitude control) to give a sense of front and back of the sound source, and the analog delay causes a time difference between the left and right sounds (phase control). It was intended to give a sense of left and right. However, in such a sound image localization method using an analog circuit, it is not easy technically and costly to accurately realize the HRTF (head related transfer function) in each sound image localization for each phase and amplitude for each frequency. Generally, it has been difficult to localize a sound source at an arbitrary position in a wide space exceeding 180 degrees.

Further, with the recent development of digital processing technology, there is a sound image localization method realized by a digital circuit, for example, there is a "sound image forming method and its device" described in Japanese Patent Laid-Open No. 2-298200. In the sound image localization method using this digital circuit, the signal from the sound source is processed by FFT (Fast Fourier Transform).
Transform) is processed and processed on the frequency axis to give a level difference and a phase difference depending on the frequency to the left and right channel signals to digitally control the localization of the sound image. The frequency-dependent level difference and phase difference at each sound image localization position of this method are collected as experimental data using an actual listener. But,
In this way, the sound image localization method using a digital circuit has the drawback that the circuit scale becomes extremely large when attempting to localize the sound image accurately and precisely, and it is only used as a special professional recording system. It was
At the recording stage, sound image localization processing (for example, flight sound movement) was performed, and the resulting sound (music) signal was recorded. The processed signal is reproduced by a normal stereo reproducing device, so that a moving effect of a sound image is generated.

By the way, recently, amusement game machines and computer terminals utilizing virtual reality have appeared. Also in these game machines and terminals, there is a demand for realistic sound image localization corresponding to the screen. For example, in game consoles, the movement of flight sounds that matches the movement of an airplane on the screen is beginning to be required. In this case, if the flight course of the airplane has been decided, the sound (music) that has been subjected to the movement processing of the sound image according to the movement is inserted in advance, and the sound (music) is simply reproduced on the game machine side, Is enough.

However, in the game machine (or terminal device), the flight course (position) of the airplane varies depending on the operation of the operator, and the sound image is adjusted in real time according to the operation of the operator. It becomes necessary to perform the moving process of and reproduce it. This point is significantly different from the sound image localization processing for records described above. For this reason, a sound image localization processing device is required for each game machine, but in the above-described conventional method, the signal from the sound source is FFT-converted, processed on the frequency axis, and inverse FFT-converted again for reproduction. Since it was necessary, the circuit scale became very large and could not be a practical solution. Further, in the above-described conventional method, since the sound image localization is based on the data on the frequency axis (data of the level difference and the phase difference depending on the frequency), if an approximation process is performed in order to reduce the circuit scale, , HRTF
The approximation of could not be performed accurately, and as a result, it was difficult to localize in a wide space beyond the range of 180 degrees.

Therefore, the present applicant has already devised a sound image localization control method as an alternative to the conventional method. This is a sound image localization control method that enables localization in a wide space exceeding the range of 180 degrees, while the circuit scale is small and cost effective. The first feature is that the sound source is generated by a pair of convolvers. The signal from is processed on the time axis to localize the sound image, and the circuit scale is made very small.
Secondly, the sound image localization processing data of the convolver is finally supplied as IR (impulse response) data on the time axis to accurately approximate the HRTF without impairing the sound image localization feeling, Scale (number of convolver coefficients)
Is made smaller.

[0007]

However, in the above-mentioned new sound image localization control method, the time response (transfer characteristic) of the convolver is obtained from the measurement result of the HRTF, but when viewed as a frequency response, it is sharp. It was a characteristic with peaks and dips. When this is used as it is as the transfer characteristic (time response) of the convolver, since unnecessary peaks and dips on the frequency characteristic are reproduced,
When implementing sound image localization, the problem of unnatural sound quality remained. This indicates that there is a limit to actual measurement. Further, the time response (impulse response) of the convolver itself does not always converge well because it reproduces sharp peaks and dips, and the problem remains that the scale (coefficient) of the convolver does not become so small.

[0008]

In order to solve the above-mentioned problems, the present invention provides a plurality of transducers (speakers sp) which are spaced apart from each other as shown in FIGS. 1, 2 and 9.
, Sp2), a plurality of signal conversion circuits (convolvers; coefficients cfLx, cfR) supplied with the same sound source (X)
A sound image localization control method for reproducing a signal processed by a convolutional arithmetic processing circuit composed of a cancel filter, which is x, and making a listener feel that a sound image is localized at an arbitrary position (x) different from the transducer. In the above, the signal conversion circuit (convolver) is based on the transfer function (HRTF; head related transfer function) measured at each sound image localization position.
The transfer characteristic (impulse response) of (step 10
1 to 104), (the obtained transfer characteristic is FFT-transformed into a discrete frequency response, and the moving average processing of the width according to the critical bandwidth is performed, and the moving-averaged value is subjected to the inverse FFT transformation. Then, there is provided a sound image localization control method characterized by using this as a transfer characteristic (impulse response) of the signal conversion circuit (step 105).

[0009]

According to the sound image localization control method as described above, the moving average is performed in the bandwidth optimized in the critical bandwidth, and as is clear from FIGS. 9C and 9D, the sound image localization is performed. Unnecessary peaks and dips are removed while leaving the necessary frequency response characteristics, and the transfer characteristic (convolver impulse response; coefficient) of the signal conversion circuit is determined based on this.

[0010]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of a sound image localization control method according to the present invention will be described below with reference to the drawings. First, the basic principle of the sound image localization control method will be described. This is a technique for localizing a sound image at an arbitrary position in space by using a pair of transducers (which will be described below by taking a speaker as an example) arranged apart from each other.

FIG. 3 is a principle diagram of sound image localization. sp1,
sp2 is a speaker arranged in front of and on the left and right of a listener (sometimes referred to as a listener in the embodiment), sp2
1 to h1L the head-related transfer characteristic (impulse response) from the listener's left ear to h1R, sp
The head-related transfer characteristics from 2 to the left and right ears are h2L and h2R. In addition, the head-related transfer characteristics to the left and right ears of the listener when the actual speaker is placed at the target localization position x are pL.
x, pRx. Here, each transfer characteristic is obtained by actually measuring with a speaker in a sound space and a microphone at both ears of a dummy head (or human head), and subjected to appropriate waveform processing.

Next, signals obtained by passing the sound source X to be localized through signal converters cfLx and cfRx (transfer characteristics by a convolver) are sp1 and sp, respectively.
Consider playing in 2. At this time, if the signals obtained in the left and right ears of the listener are eL and eR, then eL = h1L.cfLx.X + h2L.cfRx.X (Equation 1) eR = h1R.cfLx.X + h2R.cfRx.X (〃) Meanwhile, the source X Let dL and dR be the signals obtained in the left and right ears of the listener when is reproduced from the target localization position: dL = pLx · X (Equation 2) dR = pRx · X (〃)

Here, if the signals obtained in the left and right ears of the listener by reproduction of sp1 and sp2 match the signal when the source is reproduced from the target position, the listener is as if the speaker exists at the target position. The sound image will be recognized. From these conditions eL = dL, eR = dR, and (Equation 1) and (Equation 2), X is deleted and h1L · cfLx + h2L · cfRx = pLx (Equation 3) h1R · cfRx + h2R · cfRx = pRx (Equation 3) CfLx, cfRx is calculated from cfLx = (h2R · pLx−h2L · pRx) / H (equation 4a) cfRx = (− h1R · pLx + h1L · pRx) / H (〃) where H = h1L · h2R-h2L · h1R (Equation 4b) Therefore, if the signal to be localized by the convolver (convolution operation processing circuit) is processed using the transfer characteristics cfLx and cfRx calculated by (Equation 4a) and (Equation 4b), the target position x can be obtained. The sound image can be localized.

Although various concrete methods of realizing the signal conversion apparatus are conceivable, it may be realized by using an asymmetric FIR digital filter (convolver). The final transfer characteristic when used in the FIR digital filter is a time response function. That is, as the transfer characteristics cfLx and cfRx at the required localization position x, the coefficients of cfLx and cfRx are used as the coefficients for realizing the one obtained by (Expression 4a) and (Expression 4b) by one FIR filter processing. Create in advance and prepare as ROM data. R
If the coefficient of the required sound image localization position is transferred from the OM to the FIR digital filter and the signal from the sound source is subjected to the convolution calculation processing and reproduced from the pair of speakers, the sound image is localized at a desired arbitrary position.

The sound image localization control method based on the above principle will be described in detail with reference to FIG. FIG. 1 shows steps of a method for controlling a sound image localization.

Head related transfer
Function; hereinafter referred to as HRTF) measurement (step 101) This will be described with reference to FIGS. 4 and 5. Figure 4 shows HRTF
The measurement system of FIG. A pair of microphones ML and MR are installed on both ears of the dummy head (or human head) DM, and the sound recorded by the speaker SP is received, and the recorder DAT is received.
Source sound (reference data) refL, refR
And the sound to be measured (measurement data) L and R are synchronously recorded.

As the source sound XH, impulse sound, white noise, and other noises can be used. In particular, from the viewpoint of statistical processing, white noise is
Since it is a continuous sound and the energy distribution is constant over the audio band, it is possible to use SN by using white noise.
The ratio is improved. A plurality of angles θ in the space where the position of the speaker SP is 0 ° (°) on the front side (for example,
As shown in FIG. 5, it is installed at 12 points every 30 degrees, and continuously recorded for each predetermined time.

HRTF impulse response (Impulse Re
sponse; hereinafter referred to as IR) (step 10)
2) In step 101, the source sounds (reference data) refL and refR and the to-be-measured sounds (measurement data) L and R, which are synchronously recorded, are stored in a workstation (not shown).
Process above. Letting X (S) be the frequency response of the source sound (reference data), Y (S) be the frequency response of the sound to be measured (measurement data), and IR (S) be the frequency response of the HRTF at the measurement position (Equation 5). There is an input / output relationship as shown in. Y (S) = IR (S) · X (S) (Equation 5) Therefore, the frequency response of the HRTF is IR (S) = Y (S) / X (S) (Equation 6) is there.

Therefore, the frequency response X of the reference
(S), the frequency response Y (S) of the measurement data is obtained by cutting out the data obtained in step 101 with a window that is time-synchronized, and performing a finite Fourier series expansion by FFT conversion to calculate a discrete frequency, From 6), HRT
The frequency response IR (S) of F is obtained by a known calculation method. In this case, in order to improve the accuracy of IR (S) (improvement of SN ratio), it is preferable to calculate IR (S) for several hundred windows that are temporally different and average them.
Then, the frequency response IR (S) of the calculated HRTF is subjected to inverse FFT conversion to obtain a time-axis response (impulse response) IR (first IR) of the HRTF.

IR (impulse response) shaping processing (step 103) Here, the IR obtained in step 102 is shaped. First, for example, by the FFT transform, the first obtained in step 102
The IR of the is expanded at discrete frequencies over the audio spectrum, and unnecessary band (a large dip occurs in the high range, but this is unnecessary that does not affect sound image localization) is removed by a BPF (bandpass filter). To do. By limiting the band in this way, unnecessary peaks and dips on the frequency axis are removed and unnecessary coefficients do not occur in the cancel filter, so that the convergence is improved and the coefficients can be shortened.

Then, the band-limited IR (S) is inverted F
FT conversion is performed, and IR (impulse response) is cut out on the time axis by a window (for example, a cosine function window) and subjected to window processing (second IR). By performing the window processing, the effective length of the IR is not long, the convergence of the cancel filter is improved, and the sound quality is not deteriorated. FIG. 7 shows a specific example of IR (impulse response) of HRTF. The horizontal axis is time (sample clock is 4
The clock unit is 8 kHz), and the vertical axis is the amplitude level. The chain double-dashed line shows the window.

Cancellation filters cfLx, cfRx
(Step 104) The cancel filters cfLx and cfRx, which are convolvers (convolutional integration circuits), are cfLx = (h2R · pLx−h2L · pRx), as shown in the above (Equation 4a) and (Equation 4b). / H (Formula 4a) cfRx = (-h1R.pLx + h1L.pRx) / H (〃) However, it is H = h1L.h2R-h2L.h1R (Formula 4b).

Here, the speakers sp1 and sp to be arranged are
Head related transfer characteristics h1L, h1R, h2L, h2R
And, as the head-related transfer characteristics pLx and pRx when the actual speaker is arranged at the target localization position x, the shaped second IR (obtained for each angle θ obtained in steps 101 to 103 above). Impulse response). The head-related transfer characteristics h1L and h1R correspond to the position of the L-channel speaker in FIG. 6, and if they are installed at 30 degrees (θ = 330 degrees) from the front to the left, θ
= Use IR of 330 degrees. Head-related transfer characteristics h2R, h2
L corresponds to the position of the R channel speaker in the same figure, and if it is installed at 30 degrees (θ = 30 degrees) from the front to the right, an IR of θ = 30 degrees is used (that is, System for sound image reproduction (for example, shown in Fig. 2)
Choose one close to). Then, the head-related transfer characteristics pLx, p
As Rx, by substituting the IR for every 30 degrees in a wide space (entire space) beyond the range of 180 degrees 90 degrees to the left and right from the front which is the target sound source localization position, C of the whole space corresponding to it
fLx, cfRx, that is, 12 sets of cancellation filters cfLx, cfRx are obtained for every 30 degrees (in FIG. 6, a position of 240 degrees is taken as an example). The cancel filters cfLx and cfRx group are finally obtained as IR (impulse response) which is a response on the time axis.

The calculation of the cancel filters cfLx and cfRx according to (Equation 4a) is as follows. First, H ^-1 , which is a kind of inverse filter for H in (Equation 4b), is obtained by the least squares method, and this is subjected to inverse FFT conversion to obtain a time function h (t). Also, the terms h1L and h1 of (Equation 4a)
By expressing R, h2L, pRx, pLx, and h2R by a time function, respectively, the following equation holds. cfLx (t) = (h2R.pLx-h2L.pRx) .h (t) (Formula 7) cfRx (t) = (-h1R.pLx + h1L.pRx) .h (t) (〃) Therefore, these (Formula 7) The coefficients of the cancellation filters cfLx and cfRx are obtained from 7).
As is clear from (Equation 7), the cancellation filter cf
In order to shorten the coefficients of Lx and cfRx, it is extremely important to shorten each head-related transfer characteristic h1L, h1R, h2L, pRx, pLx, h2R. Therefore, as described above, various processes such as the window process and the shaping process are performed in steps 101 to 103 to shorten the head-related transfer characteristics h1L, h1R, h2L, pRx, pLx, h2R. The cancellation filter coefficient cfLx,
The concrete coefficient sequence of cfRx is shown. The horizontal axis is time (time in clock units where the sample clock is 48kHz), and the vertical axis is the amplitude level. The chain double-dashed line shows the window.
However, the coefficient (group) cf of these cancellation filters
In terms of frequency response, Lx and cfRx have unnecessary peaks and dips.

Cancel filter (convolver; in this embodiment, a final cancel filter that has undergone scaling processing, which will be described later, is called a convolver).
Moving average processing of coefficients cfLx and cfRx (step 105) Then, the coefficient cf of the cancel filter (convolver)
Lx and cfRx are FFT-transformed to obtain a frequency response, which is subjected to moving averaging in a width corresponding to the critical bandwidth. This is the main part of the present invention, and will be described below with reference to FIGS. As a technique for removing the above-mentioned unnecessary peaks and dips, cFLx and cFRx obtained by the above-mentioned (Equation 4a) and (Equation 4b) are FFT-transformed to CF.
Lx (S) and CFRx (S) are obtained, CFLx (S) and CFRx (S) obtained as the discrete frequency responses are subjected to moving averaging, and the moving averaged discrete frequency responses are subjected to inverse FFT.
It is conceivable to convert and use it as the time response of the cancellation filter.

Generally, when performing moving averaging, it is general that a certain bandwidth is set and each frequency band is moving averaged with the same bandwidth. However, human hearing has a characteristic called "critical band", which is that "the sound is discriminated and the frequency is analyzed based on the bandpass characteristics that are arranged in the entire audible frequency band, and the passband width is narrower in the lower range and wider in the higher range". There is. The present application focuses on this critical band, and adopts a method of optimizing the bandwidth for moving and averaging depending on the frequency band according to the critical band. The critical bandwidth CBc (Hz) is expressed by the following equation when expressed by the center frequency f (Hz).

[0027]

[Equation 1]

FIG. 9 shows a specific example of the above processing contents. FIG. 9A shows the time response of the cancellation filter obtained by the above-mentioned (Equation 4a) and (Equation 4b) based on the head-related transfer function obtained by measurement (at the same stage as FIG. 8). is there). The figure (B) displays the discrete frequency response and the critical bandwidth CBc obtained by the FFT transformation of the figure (A). The figure (C) shows the discrete frequency response obtained by moving and averaging the figure (B) with the critical bandwidth. FIG. 7D is the time response of the cancellation filter obtained by performing the inverse FFT conversion on FIG.

By doing so, FIGS. 9C and 9D are obtained.
As is clear from the above, it is possible to remove unnecessary peaks and dips in the high range while keeping the characteristics of the frequency response in the low range necessary for sound image localization. By the above process, deterioration of the sound quality due to unnecessary peaks and dips is suppressed while maintaining good sound image localization, and at the same time, the scale of the cancel filter (convolver) is reduced.

Scaling of the cancellation filter at each localization point x (step 106) Further, the spectrum distribution of the sound source (source sound) actually processed by the convolver (cancellation filter) is like pink noise statistically. There are things that are distributed or that drop gently in the high range.In any case, since the sound source is different from a single sound, there is a risk that overflow will occur when convolution calculation (integration) is performed and distortion will occur. is there.

Therefore, in order to prevent overflow,
Of the coefficients of the cancel filters cfLx, cfRx, the maximum gain (for example, the cancel filters cfLx, c
Find the square sum of each sampled value of fRx) and scale all the coefficients so that overflow does not occur when the coefficient and the white noise of 0 db are convoluted. In practice, it is advisable to attenuate the maximum absolute value to be about 0.1 to 0.4 (for example, 0.2) with respect to the allowable level (amplitude) 1.

Then, by using the window (cosine window) shown in FIG. 8, window processing is performed so that both ends become 0 according to the number of actual convolver coefficients, and the effective length of the coefficients is shortened. In this way, the data group (in this example, 12 sets of convolver coefficient groups capable of sound localization for every 30 degrees) cfLx and cfRx that are subjected to scaling processing and finally supplied to the convolver as coefficients are obtained.

A signal from a sound source is convolved and reproduced (step 107) For example, as a sound reproducing device of a game machine, as shown in FIG.
A pair of loudspeakers sp1 and sp2 are arranged at intervals, and an acoustic signal processed by a pair of convolvers (convolution operation processing circuits) is reproduced on the pair of loudspeakers sp1 and sp2. A signal from the same sound source X (for example, a flight sound from a game synthesizer) is supplied to the pair of convolvers, and the IR coefficients cfLx and cfR created in step 105 are supplied.
x (for example, flight sound 120 degrees to the left rear (θ = 240 degrees)
When you want to localize the sound image at the position of, the coefficient of θ = 240 degrees)
Is selected and set as the convolver. For example, based on a sound image localization command from a main CPU (central processing unit) of a game machine or the like, the control sub CPU transfers the coefficient of a desired localization position from the coefficient ROM to the pair of convolvers.

In this way, the signal from the sound source X is subjected to convolution calculation processing on the time axis by the pair of convolvers, and the pair of speakers sp1 and sp1 arranged apart from each other.
Played from 2. The sound reproduced from the pair of speakers sp1 and sp2 has its sound image localized so that the crosstalk to both ears is canceled and the sound source is located at a desired position, and is heard by the game operator (listener), and extremely. Played as a sound with a sense of reality. As for the coefficient of the convolver, the optimum sound image position is sequentially selected and switched with the transition of the movement of the airplane according to the operation of the operator. Further, when the flight sound is changed to, for example, a missile sound, the source sound from the sound source X is changed from the flight sound to the missile sound. In this way, the sound image can be freely localized at any position.

In the above description, the critical bandwidth is specifically defined as being defined by an equation, but the critical bandwidth is not limited to this. What is similar to this definitional expression, or approximated by a logarithmic expression, may be any as long as its width is narrower in the lower range and wider in the higher range. Further, as a transducer for reproduction, a headphone can be used instead of the pair of speakers sp1 and sp2. In this case, since the HRTF measurement conditions are different, it is advisable to prepare the coefficient separately and switch it according to the reproduction situation. Further, the IR (impulse response) shaping process shown in step 103 is not always necessary, and the sound image localization control is possible even if omitted. Further, the configuration described in the embodiment, in which the signal processed by the pair of convolvers to which the same sound source is supplied from the pair of transducers arranged apart from each other is reproduced, is the minimum configuration for obtaining the effect of the present application. Is shown. Therefore, if necessary,
Of course, a pair, that is, two or more transducers and convolvers may be additionally configured. Further, when the convolver coefficient is long, the coefficient may be divided into a plurality of convolvers. .

[0036]

As described above in detail, in the sound image localization control method according to the present invention, a signal processed by a plurality of signal conversion circuits to which the same sound source is supplied from a plurality of transducers arranged at a distance. In the sound image localization control method for reproducing a sound image to the listener as if the sound image is localized at an arbitrary position different from the transducer, the signal conversion circuit based on the transfer function measured at each sound image localization position. In addition to obtaining the transfer characteristic of, the moving average processing of the width corresponding to the critical bandwidth is performed on the obtained transfer characteristic, and this is used as the transfer characteristic of the signal conversion circuit. The transfer function is obtained by removing the unnecessary peaks and dips while moving averaged with the optimized bandwidth and leaving the characteristics of the frequency response necessary for sound localization.
Natural sound quality can be obtained. Also, as a result of the above moving average,
Since the convergence of the time response to be realized can be accelerated,
The size of the signal conversion circuit can be reduced, which is advantageous in terms of cost. Therefore, the sound image localization control can be easily performed by mounting on a consumer game machine, a computer terminal, or the like.

[Brief description of drawings]

FIG. 1 is a chart showing steps of a sound image localization control method according to the present invention.

FIG. 2 is a configuration diagram of a sound image localization apparatus based on a sound image localization control method according to the present invention.

FIG. 3 is a configuration diagram showing a basic principle of a sound image localization control method according to the present invention.

FIG. 4 is a configuration diagram showing an HRTF (head related transfer function) measurement system.

FIG. 5 is a diagram illustrating the points of HRTF measurement.

FIG. 6 is a diagram illustrating an example of calculating a cancel filter.

FIG. 7 is a diagram showing a specific example of IR (impulse response) of HRTF.

FIG. 8 is a diagram showing a specific example of coefficients of a cancel filter.

FIG. 9 is a diagram illustrating a sound image localization control method according to the present invention, in which FIG. 9A is a time response of a signal processing circuit (convolver) obtained based on a head related transfer function obtained by measurement; The figure (B) shows the discrete frequency response and the critical bandwidth obtained by the FFT transformation of the figure (A), and the figure (C) shows the moving average of the figure (B) with the critical bandwidth. The converted discrete frequency response is displayed, and FIG. 6D is the time response of the signal processing circuit (convolver) obtained by performing the inverse FFT conversion on FIG.

[Explanation of symbols]

101 Step of measuring head related transfer function (HRTF) 102 Step of calculating impulse response of HRTF 103 Step of shaping IR (impulse response) 104 Step of calculating cancel filters cfLx, cfRx 105 Movement of cancel filters cfLx, cfRx Averaging process step 106 Step of scaling the cancellation filter 107 Step of convoluting and reproducing a signal from a sound source sp1, sp2 Speakers h1L, h1R Speaker transfer from the speaker sp1 to the left and right ears of a listener h2L, h2R Speaker sp2 To the listener's left and right ears pLx, pRx Head to head listener's left and right ears when an actual speaker is placed at the intended localization position x cfLx, cfRx Cancellation Filter (convolver) and its coefficient DM dummy head (or human head) M listener (game operator, listener) X sound x purpose to sound image localization position CBc critical bandwidth f center frequency

Claims (1)

[Claims] 1. A signal processed by a plurality of signal conversion circuits to which the same sound source is supplied is reproduced from a plurality of transducers arranged apart from each other, and a signal is reproduced by a listener at an arbitrary position different from that of the transducer. In the sound image localization control method that makes the user feel that the sound image is localized, the transfer characteristic of the signal conversion circuit is obtained based on the transfer function measured at each sound image localization position, and the critical band for the obtained transfer characteristic is determined. A sound image localization control method characterized in that a moving average processing of a width according to a width is performed, and this is used as a transfer characteristic of the signal conversion circuit.