JP2953011B2 - Headphone sound field listening device - Google Patents

Description

Description: BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound field listening apparatus that enables a user to listen to a sound field equivalent to speaker reproduction by listening to headphones.

2. Description of the Related Art In recent years, in the audio-visual field, a technical trend is changing from a conventional stereo reproduction to a method of dynamically controlling a sound field in accordance with an image. The first prior art is disclosed in U.S. Pat. Nos. 3,674,792, 3,632,886, and U.S. Pat.
There is a Dolby Surround active matrix type sound field control device shown in 3959590.

Hereinafter, a first conventional sound field control device will be described with reference to the drawings.

First, a Dolby surround encoding method will be described.

FIG. 7 is a block diagram showing a configuration of a Dolby surround encoder.

In FIG. 7, 701 is an L (left channel) signal input terminal, 702 is an R (right channel) signal input terminal, and 703 is a C signal input terminal.
(Center channel) signal input terminal, 704 is S (surround channel) signal input terminal, 705 is 3 [dB] for C signal
An attenuator for attenuating the signal, 706 is an adder for adding the output of the attenuator 705 to the L signal, 707 is an adder for adding the output of the attenuator 705 to the R signal, 708 is an attenuator for attenuating the S signal by 3 [dB], 709 Is a band-pass filter that allows the output of the attenuator 708 to pass 100 [Hz] to 7 [kHz], 710 is a modified B-type noise reduction encoder that encodes the output of the band-pass filter 709, and 711 is a modified B-type noise reduction encoder 710. 712 is a phase shifter that generates a signal having a phase difference of ± 90 [deg] with respect to the output of the adder 712, and the adder 712 adds the +90 [deg] output of the phase shifter 711 to the output of the adder 706. The adder 714 adds the -90 [deg] output of the adder to the output of the adder 707.
Is an Lt signal output terminal that outputs the output of the adder 713 as an Lt (encoder left channel) signal, and 715 is an Rt signal output terminal that outputs the output of the adder 713 as an Rt (encoder right channel) signal.

The operation of the Dolby surround encoder configured as described above will be described.

The L (left channel) signal input to the Dolby surround encoder is a speaker arranged at the front left of the listening position in the listening room, the R (right channel) signal is a speaker arranged at the right front, and the C (center channel) signal. Are speakers arranged on the front, and the surround channel signal S is a signal mixed on the assumption that it is reproduced by two speakers arranged on the left and right behind. The C signal is attenuated by 3 [dB] in the attenuator 705, and is added to the L signal in the adder 706 and to the R signal in the adder 707, respectively. The S signal is attenuated by 3 [dB] by the attenuator 708, and further band-limited to 100 [Hz] to 7 [kHz] by the band-pass filter 709.
The output of the bandpass filter 709 is encoded by a modified B-type noise reduction encoder 710. This encoding will be described later. The output of the modified B-type noise reduction encoder 710 is shifted by +90 [deg] in the phase shifter 711, and is added to the output of the adder 706 by the adder 712. Adder
The output of 712 is the Dolby surround encoder output Lt. Similarly, the modified B-type noise reduction encoder 71
The output of 0 is shifted by -90 [deg] in the phase shifter 711,
Is added to the output of the adder 707 to become the Dolby surround encoder output Rt.

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

Lt = L + 0.7C + 0.7jS (1) Rt = R + 0.7C-0.7jS (2) where j represents (-1) ^1/2 and the phase rotation is 90 [d].
eg].

In the modified B-type noise reduction encoder 710, the amplitude frequency characteristic changes according to the level of the input signal. By decoding this encoded signal, high-frequency components of noise generated in the transmission medium can be reduced. Table 1 shows the amplitude frequency characteristics using the input signal level of the modified B-type noise reduction encoder as a parameter.

Next, a decoder of the Dolby surround active matrix type sound field control device will be described.

FIG. 8 is a block diagram showing a configuration of a decoder of the Dolby surround active matrix type sound field control device.

8, reference numeral 804 denotes an input terminal for an encoder output Lt (left channel) signal, 802 denotes an input terminal for an encoder output Rt (right channel) signal, 803 denotes an input balance control device for adjusting the balance between Lt and Rt, and 804. Is a level controller for adjusting the absolute levels of the balanced signals Lt and Rt. 805 is an L (left channel) signal, an R (right channel) signal, and a C (center channel) from the absolute level adjusted signals Lt and Rt. Adaptive matrix for generating signal, S (surround channel) signal, 806 is adaptive matrix 805
807 is a low-pass filter that passes a signal of 7 [kHz] or less of the delayed S signal, and 808 attenuates noise of a component of 7 [kHz] or less of the S signal. A modified B-type noise reduction decoder,
Reference numeral 809 denotes an L signal, an R signal, a C signal output from the adaptive matrix 805, and a modified B-type noise reduction decoder 80.
8 is a master level control device for controlling the level of the S signal output by 8; 810 is a listening room; 811 is a speaker arranged at the front right of the listening position in the listening room and reproducing the R signal output by the master level control device 809. , 812 are arranged in front of the listening position in the listening room and reproduce the C signal output by the master level control device 809. 813 is arranged in front of the listening position in the listening room to the left of the listening position. , A speaker for reproducing the L signal output by the master level control unit 809, and a speaker for reproducing the S signal output from the master level control unit 809, and 815, a speaker for reproducing the L signal output from the listening room in the listening room. A speaker that is disposed at the rear and reproduces the S signal output by the master level control device 809.

The operation of the Dolby Surround active matrix type sound field control device decoder configured as described above will be described.

The encoder outputs Lt and Rt expressed by the equations (1) and (2) are input to the Dolby surround active matrix type sound field control apparatus decoder.

The input balance control device 803 adjusts the balance between the input signals Lt and Rt. The level control device 804 adjusts the absolute levels of the input signals Lt, Rt. The adaptive matrix 805 controls four output signals of L, R, C and S according to the level difference between the input signals Lt and Rt. Therefore, it is necessary to adjust the balance and absolute level of the input signals Lt and Rt described above. The processing of the adaptive matrix 805 will be described later in detail. The delay device 806 delays the S (surround channel) signal of the adaptive matrix 805 by 15 to 30 [ms]. The low-pass filter 807 passes a signal of 7 [kHz] or less of the delayed S signal. Modified B-type noise reduction decoder 808
Reduces high frequency noise generated in the transmission medium included in the S signal. Modified B-type noise reduction decoder 80
Table 2 shows the amplitude frequency characteristics using the eight input levels as parameters. The characteristics of the decoder are opposite to the characteristics of the encoder shown in Table 1.

Master level controller 809 is adaptive matrix 8
A quadruple volume controller that controls the level of the L (left channel) signal, R (right channel) signal, C (center channel) signal output by the 05, and the S (surround channel) signal output by the modified B-type noise reduction decoder 808. is there.

R signal, C signal, L output by master level controller 809
Signal and S signal are speakers placed in the listening room
Reproduced on 811-815.

Here, the adaptive matrix 805 will be described.

FIG. 9 is a block diagram showing a configuration of the adaptive matrix 805. In FIG. 9, reference numeral 901 denotes an Lt input terminal; 902, an Rt input terminal; 903, a band-pass filter for limiting the Lt and Rt signal bands; and 904, L '(band-limited Lt).
905 is an adder that adds C and R '(band-limited Rt) to generate a C' signal, 905 is a subtractor that calculates the difference between L 'and R' and generates an S 'signal, and 906 to 909 are L. , R ', C', S 'full-wave rectifier circuit, 910 is a logarithmic difference circuit that outputs the logarithmic difference _DLR between R' and L ', 911 is the logarithm of S' and C ' logarithmic differential circuit for outputting a difference D _CS, 912 is a logarithmic differential circuit
A threshold switch for determining whether the output of 910 or 911 is within a predetermined range, and 913 is a time constant 22 according to the determination result of the threshold switch 912.
[Ms] or a bi-time constant circuit for processing the output D _LR of the logarithmic difference circuit 910 in the low-pass filter 448 [ms], the time constant 22 [m depending on 914 the determination result of the threshold switch 912
logarithmic difference circuit with low-pass filter of [s] or 448 [ms]
Twin time constant circuit for processing the output D _CS 911, 915 E _L multiplied by the coefficient corresponding to the polarity on the output of the bi-time constant circuit 913, the polarity dividing circuit for outputting a E _R, 916 is double time constant circuit A polarity division circuit that outputs E _C and E _S obtained by multiplying the output of 914 by a coefficient according to the polarity, and 917 denotes outputs E _L and E _R of the polarity division circuit 915 and outputs E _C and E _{S of the} polarity division circuit 916 by, E _LL, E _LR input signal Lt, controls the _{Rt, E RL, E RR,} E CL, E CR, E SL, a voltage controlled amplifier that outputs the E _SR, 918 is the output of the voltage controlled amplifier 917 E _LL , E _LR , E
_RL, a coupling network which E _RR, adds _{E CL, E CR, E SL} , E SR and the input signal Lt, previously constant multiple is decided Rt, and outputs L, R, C, and S.

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

The adaptive matrix 805 calculates the difference between the logarithms of the signal levels for the LR axis and the CS axis, and detects from which direction the signal is dominant based on the difference. Then, the signal in the dominant direction is output as it is, and the signal in the other direction is attenuated, thereby enhancing the sense of direction of the reproduced sound.

The bandpass filter 903 converts the input signals Lt, Rt from 100 [Hz] to
The band is limited to 7 [kHz].

The outputs L 'and R' of the bandpass filter 903 are given by the following equations (1).
As shown in equation (2), L and L of the encoder input are respectively
The R signal is the main component. The outputs of the adder 904 and the subtractor 905 are expressed by the equations (3) and (4), respectively.

C '= C + 0.7 (L + R) (3) S' =-jS + 0.7 (LR) (4) From equations (1) and (2), C 'and S' are C and S signals, respectively. Is the main component.

The L ', R', C ', and S' signals are respectively full-wave rectifier circuits 906-9.
09 is full-wave rectified. After full-wave rectification, a pair of L ', R' and C ', S' are processed by logarithmic difference circuits 910 and 911, respectively, to obtain outputs D _LR and D _CS . The processing of the logarithmic difference circuits 910 and 911 is expressed by the equations (5) and (6), respectively.

_{D LR = log a (R '} / L') ... (5) D CS = log a (S '/ C') ... (6) D LR indicates which of LR respect LR axis is dominant, D _CS Indicates which of CS is predominant with respect to the CS axis.

Threshold switch 912 is L and R or C and S
When the level difference is large, the adaptive matrix 80
Double time constant circuit 91 to quickly change the outputs L, R, C, S of 5
A short time constant of 22 [ms] of 3,914 is selected. Conversely, when the level difference is small, a long time constant of 484 [ms] is selected and L, R,
Change the C and S signals slowly.

The threshold switch 912 selects the time constant of 484 [ms] of the dual time constant circuits 913 and 914 if both the outputs D _LR and D _CS of the logarithmic difference circuits 910 and 911 are within the threshold level ± Lth. 22 if outside
Select the time constant of [ms]. Threshold switch 91
Table 3 shows the time constants ([ms]) selected by 2.

In order to realize Table 3, the threshold switch 91
Reference numeral 2 denotes four comparators for comparing the outputs D _LR and D _CS of the logarithmic difference circuits 910 and 911 with the threshold levels ± Lth, and an AND circuit for calculating the logical sum of the comparator outputs.

Twin time constant circuit 913 integrates the output D _LR of the logarithmic difference circuit 910 with a time constant of 484 or 22 [ms] in accordance with the determination result of the threshold switch 912. The integrator requires an RC integrator or one having a transient characteristic equivalent to it. Twin time constant circuit 914 is also similar processing on the output D _CS of the logarithmic difference circuit 911.

The polarity division circuit 915 is the D _LR integrated by the double time constant circuit 913.
Control voltage E for the voltage-controlled amplifier 917 at the subsequent stage according to the polarity of
_L and E _R are generated. The control voltages E _L and E _R are expressed by the equations (7) and (8).

E _L = D _LR D _LR <0 0 D _LR > 0 (7) E _R = 0 D _LR <0 -D _LR D _LR > 0 (8) Similarly, the voltage control amplifier generated by the polarity division circuit 916 9
The control voltages E _C and E _S for 17 are expressed by the equations (9) and (10).

E _C = D _CS D _CS <00 D _CS > 0 (9) E _S = 0 D _CS <0 -D _CS D _CS > 0 (10) The voltage control amplifier 917 outputs the output E _{L of the} polarity division circuit 915. , E _R and the outputs E _C , E _S of the polarity division circuit 916, the input signals Lt, Rt
By controlling the outputs _{E LL, E LR, E RL} , E RR, E CL, E CR, E SL, the E _SR. Here, to represent the input signal L _Y which is controlled by the output E _X polar division circuit and E _XY. FIG. 10 shows the relationship between the control voltage of the voltage controlled amplifier and the amplification factor.

The coupling network 918 is the output E _LL of the voltage controlled amplifier 917,
E _LR , E _RL , E _RR , E _CL , E _CR , E _SL , E _SR and the input signals Lt, Rt
The signals are added at the ratios shown in the table, and the L, R, C, and S signals are output.

In the output stage of the adaptive matrix 805, separation between channels of 25 [dB] or more can be secured.

As described above, the sound field control device decoder of the Dolby surround active matrix system detects in which direction the signal is dominant for the LR axis or the CS axis. When a predominant direction is detected, a signal in that direction is output as it is, and signals in other directions are attenuated to emphasize the sense of direction of the reproduced sound. Therefore, a sound with a clear direction, such as a line, can have a clear sense of direction. On the other hand, when the dominant direction is not detected, a normal stereo feeling is obtained.

 The above is the description of the first related art.

Also, in the acoustic field, technical trends are changing from original sound reproduction to original sound field reproduction, and sound field control devices for reproducing sound fields such as concert halls have been developed.

A second conventional technique is disclosed in, for example, Japanese Patent Application Laid-Open No. 61-257099.
There is an acoustic control device disclosed in Japanese Unexamined Patent Publication (Kokai) No. H10-26095.

Hereinafter, a second conventional sound field control device will be described with reference to the drawings.

FIG. 11 is a block diagram showing a configuration of a conventional sound field control device. In FIG. 11, reference numeral 1101 denotes an audio signal input terminal for inputting a source signal; 1102, reflected sound parameter storage means for storing parameters of reflected sound; 1103, based on each reflected sound parameter stored in the reflected sound parameter storage means 1102. A reflected sound generating means for adding a reflected sound to the source signal and outputting the reflected sound; 1104, 1105, 1106, 1107, a reflected sound generating means
A speaker that reproduces the signal processed in 1103, and 1108 is a listening room that reproduces the signal.

The operation of the conventional sound field control device configured as described above will be described below.

The audio signal input from the audio signal input terminal 1101 is reflected sound added by the reflected sound generation means 1103 based on each reflected sound parameter stored in the reflected sound parameter storage means 1102. Details of the processing will be described later. The signal processed by the reflected sound generation unit 1103 is reproduced by speakers 1103, 1104, 1105, and 1106 arranged at predetermined positions in the listening room 1107. The listener can listen to the sound field reproduced at a predetermined position in the listening room 1107.

The audio signal input from the audio signal input terminal 1101 normally has two channels. In this conventional example, the signal processed by the reflected sound generation unit 1103 has four channels. However, the number of output channels may be four or more or less depending on the processing.

The processing performed by the reflected sound generation unit 1103 will be described. Assuming that the input signal is x ₁ , x ₂ and the output signal is y ₁ , y ₂ , y ₃ , y ₄ , the input signal and the output signal have the relationship of equation (11).

Y = H * X (11) where: Y = (y ₁ , y ₂ , y ₃ , y ₄ ) ^T (12) X = (x ₁ , x ₂ ) ^T (13) H = (H _{1, H 2) ... (14} ) H 1 = (h 11, h 12, h 13, h 14) T ... (15) H 2 = (h 21, h 22, h 23, h 24) T ... (16 However, “*” indicates convolution, “ ^T ” indicates transpose,
_{Hji in} Equations (11) and (16) indicates an impulse response convolving with the i-channel signal output to the j-channel. The convolved impulse response _hji is stored in the reflected sound parameter storage means 1102. The impulse response is determined by measurement in an actual concert hall or by computer simulation based on the shape of the concert hall. In addition, in the case of realizing with hardware, an enormous scale of hardware is required to convolve all the impulse responses. For this reason, for example, _hji is divided into early reflection sound and reverberation sound, and the early reflection sound is
Reverberation to IIR all-pass reverberation generator
In some cases, processing is performed on small-scale hardware by assigning them to respective units.

 The above is the description of the second related art.

Problems to be Solved by the Invention However, the conventional configuration is based on speaker reproduction. Therefore, if a listening room having a sufficient size cannot be prepared, the sound field control device cannot be installed. Also,
Considering nearby noises, such as late at night, there is a problem that low volume reproduction has to be performed.

The present invention has been made in view of the problems of such a conventional sound field control device, and has an object to enable a sound field equivalent to a sound field reproduced by a speaker to be heard while listening to headphones. Technical issues.

Means for Solving the Problems To solve the above problems, a headphone sound field listening device of the present invention includes a sound field control signal processing device that performs a sound field control process on the basis of speaker reproduction for an input signal. A head-related transfer function compensator for compensating the output of the sound field control signal processor with a head-related transfer function, and headphones for reproducing the output of the head-related transfer function compensator.

Function The headphone sound field listening device of the present invention performs the same sound field control processing as the reproduction by the speaker similar to the conventional example in the sound field control signal processing device by the above configuration, and the head-related transfer function compensating device outputs the sound from the speakers. By compensating for the transfer function up to the ear canal entrance of the listener, the same sound pressure as at the time of speaker reproduction is generated at the ear canal entrance when listening to headphones.

Embodiment Hereinafter, an embodiment of a headphone sound field listening device of the present invention will be described with reference to the drawings.

FIG. 1 is a block diagram showing the configuration of an embodiment of a headphone sound field listening device of the present invention.

In FIG. 1, reference numeral 101 denotes a signal input terminal, 102 denotes a sound field control signal processing device that performs a sound field control process on the basis of speaker reproduction on an input signal, and 103 denotes a plurality of sound field control signal processing devices. A head-related transfer function compensator 104 for compensating for a signal to be output to the speaker using a head-related transfer function, and 104 is a headphone.

The operation of the headphone sound field listening device of the present invention configured as described above will be described below.

The sound field control signal processing device 102 is an input terminal of an encoder output Lt (left channel) signal and an encoder output Rt (right channel) among the decoders of the Dolby surround active matrix type sound field control device shown in the first conventional example. The same processing as that from the signal input terminal to the master level control device is performed, or the same processing as the processing by the reflected sound parameter storage means and the reflected sound generation means shown in the second embodiment I do.

Here, for the following description, each speaker of the Dolby surround active matrix type sound field control device shown in the first conventional example is numbered as follows. One loudspeaker is located at the left front of the listening position in the listening room and reproduces the L signal, two loudspeakers are located in front of the listening position and reproduce the C signal, three loudspeakers are disposed at the right front and reproduce the R signal, and the right rear. The speaker which reproduces the S signal and which is arranged at the rear left is 4 and the speaker which reproduces the S signal and which is disposed at the rear left is 5. In the following description, the speaker of the Dolby surround active matrix type sound field control device shown in the first conventional example will be described as an example. In the case of the sound field control device shown in the second conventional example, it is not particularly necessary to limit the number of speakers, but it is sufficient to similarly number each speaker.

Next, the processing contents of the head-related transfer function compensator 103 will be described.

In the processing of the sound field control signal processing device 102, the positions of each speaker and the listener are assumed. In this positional relationship, the transfer functions from the input terminal of the i-th (i = 1 to 5) speaker to the left and right ear canal entrances of the listener are _HiL and _HiR (i =
1 to 5). At this time, the signal S _i is output from the i-th speaker.
When (i = 1 to 5) is reproduced, the sound pressures P expressed by the equations (7) and (8) are respectively applied to the left and right ear canal entrances of the listener.
_iL ^SP and P _iR ^SP occur. This relationship is shown in FIG.

^{_{P iL SP = H iL · S}} i ... (17) P iL SP = H iR · S i ... (18) Further, the signal applied to the input terminals of the left and right headphone S _L, and S _R, received from the input terminal each transfer function up to the ear canal entrance of the left and right listener G _L, and G _R. At this time, sound pressures P _L ^HP and P _R ^HP expressed by equations (19) and (20) are generated at the left and right ear canal entrances of the listener. This relationship is shown in FIG.

P _L ^HP = G _L · S _L ... (19) P _R ^HP = G _R · S _R ... (20) Therefore, in order to obtain the same sound pressure as that reproduced by the i-th speaker by headphone reproduction, the headphone The signals S _iL , given by equations (21) and (22),
S _iR (i = 1 to 5) may be input.

S _iL = _GL ⁻¹ · H _iL · S _i ... (21) S _iR = G _R ^-1 · H _iR · S _i ... (22) Here, G _L ^-1 and G _R ^-1 are headphone inputs. This is the inverse characteristic of the transfer function from the terminal to the left and right ear canal entrances of the listener.
The inverse characteristics will be described later.

The transfer function from the input terminal of the speaker to the left and right ear canal entrances of the listener and the transfer function from the headphone input terminal to the left and right ear canal entrances of the listener will be described. The impulse response from the input terminal of the i-th speaker to the entrance of the ear canal of the listener can be measured using the speaker and the probe microphone in the anechoic room. The transfer function H used for the impulse response of the i-th speaker in equations (21) and (22)
_iL and _HiR . Further, the transfer function G _L from the headphone input terminal to the ear canal entrance of the listener, G _R is also obtained from the measurement in the same manner, the reverse characteristic from the transfer function G _L ^-1, can be calculated G _R ^-1. FIG. 4 is a waveform diagram showing a measurement example of an impulse response from the input terminal of the speaker to the entrance of the ear canal of the listener.
The figure shows a waveform diagram illustrating an example of the inverse characteristic of the transfer function from the headphone input terminal to the listener's ear canal entrance. The sampling frequency in FIGS. 4 and 5 is 50 [kHz], and the number of sampling points is 512.

Here, a method of calculating the inverse characteristic will be described. The inverse characteristic is a characteristic in which an impulse response of the entire system is used as a unit sample by connecting in series to a system having a transfer function from a headphone input terminal to a listener's ear canal entrance. There are two types of actually used inverse characteristics, one obtained from the discrete Fourier transform and the other obtained by deconvolution.
Since the inverse characteristic obtained from the discrete Fourier transform has a very long time, the present embodiment employs the inverse characteristic by deconvolution.

The impulse response from the headphone input terminal to the listener's ear canal entrance is g (k), the impulse response of the inverse characteristic to be obtained is g ⁻¹ (k), the unit sample is δ (k), and the number of samples of the inverse characteristic is m. In this case, a straight-line convolution relationship of equation (23) is established between them.

In the equation (23), n ₀ represents a time lag between input and output of the opposite characteristic, and is called a spike point.

Now, since the number of samples of the inverse characteristic is limited to m, the left side of Expression (23) does not accurately become a unit sample. Therefore, it becomes an impossible simultaneous linear equation in which the equality of the equation (23) does not hold. Therefore, the difference between the left side of equation (23) and the unit sample is set as a residual, and the inverse characteristic is estimated by minimizing the sum of squares of the residual. That is, the inverse characteristic can be calculated by solving the equation (23) as a simultaneous linear equation using the least squares method.

In the above description, the output processing for the i-th speaker has been described. However, in the processing of the sound field control signal processing device 102, processing is performed on the assumption that a total of five speakers are used. Therefore, the signals S _L and S _R given by Expressions (24) and (25) are input to the input terminals of the headphones.

As described above, the head-related transfer function compensator 103 is (24)
What is necessary is just to make a filter which performs a formula and a formula (25).

FIG. 6 shows a signal processing flow of the head-related transfer function compensator 103. In the sixth figure, 601 to 603 input terminals of the signal S _i, which is processed by the sound field control signal processor 102, 609 to 611 is transmitted from the input terminal of the i-th speaker respectively to the ear canal entrance of the left of the listener FIR filters for convolving the function _HiL (i = 1 to 5) into the signal S _i processed assuming the i-th speaker, 612 to 614 respectively denote the right ear canal entrance of the listener from the input terminal of the i-th speaker Transfer function H _iR (i
= 1 to 5) into a signal S _i processed assuming the i-th loudspeaker, 615 denotes FIR filters 609 to 61
Adder for adding the output of 1; 616 are FIR filters 612 to 614
617 is an FIR filter that convolves the output of the adder 615 with the inverse characteristic G _L ^-1 of the transfer function from the headphone input terminal to the left ear canal entrance of the listener, and 618 is an adder
An FIR filter that convolves the output of 616 with the inverse characteristic G _R ^-1 of the transfer function from the headphone input terminal to the right ear canal entrance of the listener, 619 is an output terminal of the output signal of the FIR filter 617, 620
Is an output terminal of the output signal of the FIR filter 618. The FIR filter 609 includes a shift register 604 that shifts an input signal in synchronization with a sampling frequency, and a multiplier 605 that multiplies each tap output of the shift register 604 by a filter coefficient.
To 607, and an adder 608 for adding the outputs of the multipliers 605 to 607. The output of the adder 608 is the output of the FIR filter 609.

The filter coefficient multiplied by the multipliers 605 to 607 is a transfer function H _1L from the input terminal of the first speaker to the left ear canal entrance of the listener on the time axis h _1L (k) (k = 1, 2, … M;
m is the number of samples, m = 512 in this embodiment), and the multiplier 605
Is h _1L (1), multiplier 606 is h _1L (2) _,.
07 multiplies h _1L (512). Other FIR filters have the same configuration.

The head-related transfer function compensator 103 configured as described above performs signal processing given by equations (24) and (25),
Outputs S _L and S _R.

The headphones 104 reproduce the signals S _L and S _R processed by the head-related transfer function compensator 103.

The above is the operation of the headphone sound field listening device according to the embodiment of the present invention.

This embodiment describes the speaker arrangement of the Dolby Surround active matrix type sound field control device of the first conventional example as rejected in the middle of the embodiment. Similar processing can be performed for the speaker arrangement of the sound field control device of the second conventional example.

Effect of the Invention As described above, the headphone sound field listening apparatus of the present invention performs compensation processing on the output of the sound field control signal processing apparatus by the head-related transfer function compensating apparatus. The effect of being able to hear the same sound field as in the case where it is performed is obtained. For this reason, the architectural restriction of the listening room is eliminated, and there is no need to consider nearby noise.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration of a headphone sound field listening apparatus according to an embodiment of the present invention. FIG. 2 is a plan view showing a state of transmission when a signal _Si is reproduced by an i-th speaker. FIG. 4 is a front view showing a state of transmission when a signal S is reproduced by headphones, FIG. 4 is a waveform diagram showing a measurement example of a transfer function from an input terminal of a speaker to an entrance of an ear canal of a listener, and FIG. FIG. 6 is a waveform diagram showing an example of an inverse characteristic of a transfer function from an input terminal to a listener's ear canal entrance, FIG. 6 is a block diagram showing a signal processing flow of a head-related transfer function compensator, and FIG. 7 is a first conventional example FIG. 8 is a block diagram showing the configuration of a Dolby Surround encoder in the conventional example, FIG. 8 is a block diagram showing the configuration of a decoder of the Dolby Surround active matrix type sound field control device in the conventional example, and FIG. FIG. 10 is a block diagram showing a configuration of an active matrix, FIG. 10 is a characteristic diagram showing a relationship between a control voltage and an amplification factor of a voltage controlled amplifier in the conventional example, and FIG. 11 is a configuration of a sound field control device in a second conventional example. FIG. 101 ... signal input terminal, 102 ... sound field control signal processing device,
103 ... head-related transfer function compensator, 104 ... headphone, 60
1 to 603: Input terminal, 604: Shift register, 605 to 60
7… Multiplier, 608,615,616 …… Adder, 609-614,617,6
18… FIR filter, 619,620 …… Output terminal.

Claims (3)

(57) [Claims] 1. A sound field control signal processing device for performing a sound field control process on a premise of speaker reproduction for an input signal, and a signal output by the sound field control signal processing device for reproduction by a speaker. Head transfer obtained by convolution of the sum of transfer functions from the input terminals of each speaker to the left and right ear canal entrances of the listener and the inverse characteristic of the transfer function from the headphone input terminal to the left and right ear canal entrances A headphone sound field listening device comprising: a head-related transfer function compensator that performs compensation by a function; and headphones that reproduce the output of the head-related transfer function compensator. 2. A sound field control signal processing device, comprising: an input balance control device that inputs two-channel signals and adjusts the balance between them; a level control device that adjusts an absolute level of an output of the balance control device; Based on the output of the level control device, it detects which of the signals from the front left, front, right front, and rear of the listening position is dominant, and outputs the signal in the dominant direction largely, and outputs signals in other directions. An adaptive matrix for producing a left channel signal, a right channel signal, a center channel signal, and a surround channel signal so as to attenuate the surround channel signal; a delay device for delaying a surround channel signal output by the adaptive matrix; A low-pass filter that passes a signal of 7 kHz or less; and a noise output from the low-pass filter. Decay is thereby deformed B
2. A noise reduction decoder, comprising: a master level control device for controlling a level of a left channel signal, a right channel signal, a center channel output by the adaptive matrix, and a signal output by the modified B-type noise reduction decoder.
The headphone sound field listening device according to the above. 3. A sound field control signal processing device, comprising: a plurality of speakers arranged around a listening point based on data of a reflected sound in an acoustic space, the reflected sound in the acoustic space or a model space similar to the acoustic space; In order to reproduce, the reflected sound parameter storage means for storing parameters of the reflected sound to be emitted by the plurality of speakers, and the reflected sound of the source signal based on each of the reflected sound parameters stored in the acoustic sound parameter storage means. 2. The headphone sound field listening device according to claim 1, further comprising: a reflected sound generating means for generating and outputting.