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JP2002078069

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DESCRIPTION JP2002078069
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to
characteristics inherent to a speaker by correcting the sound pressure from the speaker observed
at one listening point in an arbitrary sound field and the phase / frequency characteristics to
desired characteristics. The present invention relates to an acoustic characteristic correction
device that prevents the deterioration of sound quality due to the characteristic inherent to the
sound field and enables high-quality reproduction of sound signals such as voice and musical
sound at the listening point.
[0002]
2. Description of the Related Art A prior art (Japanese Patent Application No. 10-059703) will be
described with reference to FIG. The initial value of the transfer function C (z) of the convolver 1
is zero. The input signal s (ω) (ω: discrete frequency) is processed by the delay unit 11 and the
convolver 1 connected in parallel to this, and is sent to the speaker 2 and the reference signal
generation filter 4 as the signal x (ω). It is input. The output signal y (ω) of the microphone 3 is
input to a second delay unit 12 having the same delay time as the first delay unit 11 and an
adaptive filter 6 connected in parallel thereto. The adaptive filter 6 also receives the noise u (ω)
(¦ u (ω) ¦ 2 << ¦ y (ω) ¦ 2) supplied from the noise generator 9. The adaptive filter 6 generates a
replica c (ω) of the output signal d (ω) of the reference signal generation filter 4 from the output
r (ω) of the second delay unit 12.
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[0003]
The transfer functions of the delay units 11 and 12 are zm (m: discrete time), the transfer
function between the speaker 2 and the microphone 3 is z-tG (z), and the transfer function of the
reference signal generation filter 4 is Assuming that z− (m + t) R (z), the transfer function H (z)
of the adaptive filter 6 can be expressed as error power ωω ¦ e (ω) ¦ 2 = Σω ¦ d (ω) −c (ω)
It converges to the next value by the algorithm which makes "minimization of ¦ 2" as the teaching
principle. H (z) ≒ -z-m + z- (m + t) R (z) / z-tG (z) =-z-m + z-mR (z) / G (z) (1) this transfer function
H ( By setting z) to the convolver 1, the output signal y (ω) of the microphone 3 is controlled as
follows.
[0004]
Y (ω) = f [z-tG (z)] x (ω) = f [z-tG (z)] f [Z-m + C (z)] s (ω) = f [z-tG (Z) {z-m + H (z)}] s (ω) = f [z-(t +
m) R (z)] s (ω) (2) where f [·]: discrete Fourier transform The output signal y (ω) is a signal
obtained by correcting the input signal s (ω) to the desired amplitude and phase / frequency
characteristics given to the reference signal generation filter 4.
[0005]
In order to simplify and clarify the explanation, only the input signal s (ω0) corresponding to the
frequency ω0 is considered, and this input signal changes as follows with time k Assume that
k ≦ k1: s (ω0) ≠ 0k1 <k ≦ k2: s (ω0) = 0 k2 <k: s (ω0) ≠ 0 (a) k ≦ k1 As described above, the
adaptive filter 6 corresponding to the frequency ω0 The transfer function H (.omega.0) (= C
(.omega.0), C (.omega.0): transfer function of the convolver 1 corresponding to the frequency
.omega.0) converges to the optimal value shown in equation (1). Let this value be HOPT ≠ 0. (B)
When k1 <k ≦ k2, the output signal d (ω0) of the reference signal generation filter 4, the output
signal y (ω0) of the microphone 3, and the output signal r (ω0) of the second delay unit 12 are
all zero. Become. To minimize ¦ e (ω0) ¦ 2 = ¦ d (ω0) −C (ω0) ¦ 2, that is, ¦ e (ω0) ¦ 2 = ¦ 0− (0
+ u (ω0) H (ω0)) ¦ 2. Since the convolution noise u (ω0) is nonzero, the transfer function H
(ω0) of the adaptive filter 6 converges to zero. (C) When k2 <k, the transfer function H (ω0) of
the adaptive filter 6 converges again to the optimum value HOPT as in the case of (a).
[0006]
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That is, in the prior art, when the input signal s (ω0) fluctuates and becomes zero repeatedly, the
transfer function H (ω0) of the adaptive filter 6 which corrects the input signal s (ω0) to a
desired characteristic according to the fluctuation. ) (= C (ω0)) swings between the optimum
value HOPT and zero, and s (ω0) can not be continuously corrected to the desired characteristic.
An object of the present invention is to provide an acoustic characteristic correction device
capable of continuously correcting s (ω0) to a desired amplitude and phase / frequency
characteristics regardless of the fluctuation of s (ω0) as described above. is there.
[0007]
SUMMARY OF THE INVENTION In order to achieve the above object, according to the present
invention as set forth in claim 1, a second convolver having the same transfer function as
convolver 1 is obtained by removing the conventional second delay 12. , And only the output side
is connected in parallel with the adaptive filter 6, and an antiphase signal of the pseudo noise
generated by the noise generating unit 7 is input to the second convolver. Further, the noise
generation unit 7 analyzes the reference signal d (ω), identifies a frequency ω0 (ω0 is not
limited to one) at which this signal becomes zero, and generates pseudo noise u (ω0) of this
frequency. . For this frequency ω0 where the input signal becomes zero and hence the reference
signal becomes zero, the error power ¦ e (ω0) ¦ 2 = ¦ H (ω0) u (ω0) −C (ω0) u (ω0) The
transfer function H (ω0) of the adaptive filter 6 is constrained to the optimum value C (ω0)
which has already been set in the second convolver, by an algorithm based on the principle of
minimization of ¦ 2 . Also, the conventional first delay unit 11 becomes unnecessary.
[0008]
BEST MODE FOR CARRYING OUT THE INVENTION FIG. 1 shows an embodiment of an acoustic
characteristic correction device proposed in claim 1 of the present invention. The parts
corresponding to those in FIG. 2 are given the same reference numerals. The input signal s (ω) is
supplied to the speaker 2 and the reference signal generation filter 4 only through the convolver
1. Further, a convolver 5 having the same transfer function as the convolver 1 is provided, and
the output of the convolver 5 is added to the output of the adaptive filter 6 to be a replica c (ω).
The pseudo noise u (ω) from the noise generator 7 is input to the convolver 5 through the phase
inverter 10. In the noise generation unit 7, the noise n (ω) from the pseudo noise generator 9 is
given a weight α (ω) by the multiplier 8, and pseudo noise u (ω) = α (ω) · n (ω) It is output.
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[0009]
The initial value of the transfer function C (z) of the convolvers 1 and 5 is the m sample delay zm. The input signal s (ω) is processed by the convolver 1 and is input to the speaker 2 and the
reference signal generation filter 4 as a signal x (ω). A desired characteristic z− (t + m) R (z)
delayed by t + m samples is set as the transfer characteristic of the reference signal generation
filter 4. This delay is to stably converge the transfer function H (z) of the adaptive filter 6. The
output signal y (ω) of the microphone 3 is added to the pseudo noise u (ω) generated by the
noise generator 7 and input to the adaptive filter 6. The pseudo noise u (ω) becomes an
antiphase signal in the phase inverter 10 and is input to the second convolver 5. The noise
generation unit 7 analyzes the output d (ω) of the reference signal generation filter 4 by the
frequency weight calculation unit 13, calculates an appropriate frequency weight α (ω)
according to the result, and generates the pseudo noise generator 9. The noise n (ω) is
multiplied by the frequency weight α (ω) by the multiplier 8 to obtain the next pseudo noise u
(ω) = α (ω) n (ω). The noise n (ω) is assumed to have approximately constant intensity over all
frequencies. The frequency weight α (ω) is given, for example, as follows.
[0010]
When .vertline.d (.omega.0) .vertline.2 / .SIGMA..omega..vertline.d (.omega.).
Vertline.2.ltoreq..beta., .Alpha. (. Omega.0) = 1 (3a) .vertline.d (.omega.0) .vertline.2 /
.sigma..omega..vertline.d (.omega.). Vertline.2 When β>, 0 <α (ω0) << 1 (3b) where ¦ n (ω) ¦ 2
≒ ¦ y (ω) ¦ 2 (ω ≠ ω0) β: positive constant (eg, 10− 4) The transfer function H (z) of the
adaptive filter 6 is derived from an algorithm based on the teaching principle of error power
ωω ¦ e (ω) ¦ 2 = Σω ¦ d (ω) −c (ω) ¦ 2 minimization . Desired. When this algorithm
converges, the transfer functions C (z) of convolvers 1 and 5 are replaced by H (z). Hereinafter,
the input signal is changed in the same manner as described in the section "Problems to be
solved by the invention", that is, the input signal s (ω0) corresponding to the frequency ω0 is
considered, and the problems of the prior art are solved. Make sure. The input signal s (ω0)
changes as follows with time k.
[0011]
K ≦ k1: s (ω0) ≠ 0k1 <k ≦ k2: s (ω0) = 0 k2 <k: s (ω0) ≠ 0 (a) k ≦ k1 In a state where the
equation (3b) is satisfied Transfer function H (ω0) (= C (ω0), C (ω0): convolver 1 and adaptive
filter 6 corresponding to frequency ω0, such that ¦ u (ω0) ¦ 2 << ¦ y (ω0) ¦ 2. The transfer
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function of 5 converges to the optimum value f [Z−mR (z) / G (z)] (ω = ω0). Let this value be
HOPT ≠ 0. (B) When k1 <k ≦ k2, the output signal d (ω0) of the reference signal generation
filter 4 and the output signal y (ω0) of the microphone 3 become zero. In addition, pseudo noise
u (ω 0) of sufficient size is generated in the noise generating unit 7 in a state satisfying equation
(3a), as described in the section Means for Solving the Problems . The transfer function H
(ω0) is constrained to C (ω0) = HOPT by minimization of ω0) ¦ 2 = ¦ H (ω0) u (ω0) −C (ω0) u
(ω0) ¦ 2. (C) When k2 <k, equation (3b) is satisfied, and the transfer function H (ω0) converges
to HOPT ≠ 0 as in (a) k ≦ k1, but H (() Since ω0) = C (ω0) is already HOPT, H (ω0) remains at
the optimum value HOPT as long as the transfer function z-tG (z) between the speaker 2 and the
microphone 3 does not change.
[0012]
That is, according to the present invention, s (ω0) can be continuously corrected to the desired
amplitude and phase / frequency characteristics regardless of the variation of s (ω0) as
described above.
[0013]
As described above, according to the present invention, the first convolver having an
appropriate transfer function is used to generate the speaker input signal, and the same transfer
function as the first convolver is used. [2] Repeating a series of processing setting the transfer
function to the convolver by obtaining a transfer function for correcting the sound signal
characteristic to be reproduced to the speaker to a desired characteristic by an adaptive filter
whose output side is connected in parallel to the convolver Therefore, even in the case of
reproducing a signal in which a supplied frequency component fluctuates with time, such as
voice or musical sound, the characteristics (sound pressure, phase, and frequency) of the speaker
reproduction sound observed at one listening point in the sound field The characteristic) can
continue to be corrected to the desired characteristic.
[0014]
Brief description of the drawings
[0015]
1 is a block diagram showing the configuration of the acoustic characteristic correction device
according to an embodiment of the present invention.
[0016]
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2 is a block diagram for explaining the prior art.
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