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JPH08110783

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DESCRIPTION JPH08110783
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an
audio signal transmission circuit, and more specifically, to an audio signal transmission circuit
which corrects the response characteristic of the speaker of a high fidelity audio apparatus and
reproduces faithful source sound quality. The present invention relates to a convolver coefficient
calculation device for obtaining convolver coefficients provided in an audio signal transmission
system.
[0002]
2. Description of the Related Art Heretofore, various methods have been addressed as audio
signal transmission circuits for reproducing faithful source sound quality. For example, Japanese
Patent Application Laid-Open No. 61-195099 discloses that "the sound reproduction system best
in terms of hearing can be obtained by calculating the weight coefficient of the convolver by
making correction on the sense of hearing from the practical aspect". This is to freely set the
position of the sound image or to set the sound field and timbre (equalizer) by providing the
convolver in the transmission path, but the method of correcting the response characteristic of
the speaker is mentioned. Absent.
[0003]
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1
Further, in Japanese Patent Application Laid-Open No. 63-281510, “The inverse Fourier
transform of the transfer function having the amplitude frequency characteristic and the phase
frequency characteristic is performed to calculate the impulse response, and the obtained filter
coefficient is used as a transversal filter. By setting, desired characteristics are obtained. This is
intended to correct the response characteristics of the speaker, but the purpose is to improve
calculation when calculating the impulse response by inverse Fourier transform, and it is
intended to simply correct the characteristics of the speaker flat. It is a thing.
[0004]
Further, according to the conventional example of Japanese Patent Laid-Open No. 2-272819,
when correcting group delay frequency characteristics of a speaker with an IIR type filter flat,
transient response calculation means and frequency components with long transient response
time are A technique is disclosed to reduce the interaction between frequencies in the transient
response by providing a detection means for detecting and performing group delay correction in
addition to the frequency components to eliminate frequency components having a relatively
long response time. However, this corresponds to each in-band, and actively removes the
characteristic at a specific frequency in the band from the correction target, and reduces the
interaction specific to the IIR filter.
[0005]
By the way, in the above-mentioned Japanese Patent Application Laid-Open No. 61-195099,
there is no mention of a method of correcting the response characteristic of the speaker, and
both of the amplitude characteristic and the phase characteristic are flat. By performing
measurement and analysis so as to be (flat), for example, for correction at high frequencies, the
characteristic of the alias removal filter of the D / A converter used in the measurement system is
included in the measurement data. The characteristic is that the portion of the high region is
rapidly lifted up, and if you try to listen to what was actually corrected, it is clear that no
correction of distortion is clearly felt, but rather a sound quality with a lot of distortion.
[0006]
Further, in Japanese Patent Application Laid-Open No. 63-281510, in order to keep the
frequency flat in the low frequency region up to the operating limit of the speaker, the distortion
may be heard in the low frequency region.
[0007]
Furthermore, according to the prior art of Japanese Patent Application Laid-Open No. 2-272819,
although it is possible to detect the correction limit of each band in the band and minimize the
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2
correction error, the speaker in the low frequency region peculiar to the speaker can be
minimized. No measures have been considered for increasing the distortion in the low band
(band: not a spot) by keeping the frequency flat below the operation limit.
[0008]
The present invention has been made to solve the problems of the prior art, and eliminates
distortion of the sound image that can not be avoided by the speakers and headphones, and
simultaneously corrects the amplitude and phase to obtain a predetermined transmission
characteristic. A convolver coefficient for determining a convolver coefficient provided in an
audio signal transmission circuit and an audio signal transmission system which can enjoy
natural audio signals by maintaining only a mid frequency band constant and reproducing a
faithful original sound quality It aims at providing an arithmetic unit.
[0009]
SUMMARY OF THE INVENTION In order to achieve the above object, an audio signal
transmission circuit according to the present invention performs convolution operation
processing on a signal from a sound source according to a set coefficient and outputs the signal.
And a predetermined measurement system including a transducer, the audio signal from the
predetermined sound source is given to the transducer, and the measurement is performed based
on the impulse response waveform h (t) previously measured at the measurement position near
the transducer The impulse response f0 (t) of the specific characteristic of the impulse response
waveform h (t) at the position becomes flat only in a predetermined mid region, and the impulse
response waveform in one or both of the higher and lower bands than the mid region A
coefficient that is processed so as to be substantially the same as the characteristics of h (t) and
that supplies the calculated inverse filter coefficient to the convolver And supply means.
[0010]
Further, according to the convolver coefficient calculation device of the present invention,
measurement signal generating means for generating a measurement signal to be sent to an
audio signal transmission system, and an amplitude characteristic which is a response of the
audio signal transmission system based on the measurement signal. Among the response
characteristics obtained by the response characteristics measuring means to be found and the
response characteristics obtained by the response characteristics measuring means, the target
characteristics in which the amplitude of only the predetermined middle frequency band is
replaced flatly are determined, and the response of the audio signal transmission system is And
calculating means for determining a filter coefficient of the convolver provided in the audio
signal transmission system so as to converge on the target characteristic.
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[0011]
Further, while the target characteristic in which the amplitude of only the predetermined middle
frequency band is flatly replaced by the arithmetic means is regarded as the first target
characteristic, a band lower than the predetermined middle frequency band and In the high band,
the second target characteristic is determined by rolling off the response characteristic amplitude
at a predetermined rate, and the first and second filters of the convolver corresponding to the
first and second target characteristics. The first and second amplitude characteristics corrected
by respectively calculating the coefficients and performing the characteristic measurement based
on the setting of the first and second filter coefficients are determined, and the first and second
amplitude characteristics are calculated. The peaks of the deviation values are respectively
searched among the above, and the determined peaks of the deviation values are compared and
it is determined to select the filter coefficient with the smaller deviation. A.
[0012]
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS First Embodiment FIG. 1 is a block
diagram showing an embodiment of an audio signal transmission circuit according to the present
invention.
In FIG. 1, two-channel stereo sound sources 1L and 1R are predetermined audio signal sources.
Convolvers 2L and 2R and amplifiers 3L and 3R are provided between the speakers 4L and 4R
and the sound sources 1L and 1R for correcting the response characteristics of the speakers to
simultaneously correct the amplitude and phase characteristics.
15 switches and controls a switch to be described later and measures the response
characteristics of the speakers 4L and 4R to obtain a correction filter coefficient of the convolver
and gives the obtained correction filter coefficient to the convolvers 2L and 2R to perform
convolution operation It is a control unit that performs control to correct the speaker.
A memory 16 is provided to store correction filter coefficients obtained based on the
measurement of the response characteristics of the speakers 4L and 4R, and an audio signal is
provided when the response characteristics of the speakers 4L and 4R are measured based on
the control of the control unit 15. A switch 7L, 7R is provided to separate the convolvers 2L, 2R
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from the transmission line of the transmission line, and to provide the convolvers 2L, 2R in the
transmission line of the audio signal when correcting the response characteristics of the speakers
4L, 4R. .
[0013]
That is, the configuration shown in FIG. 1 measures the impulse response of the speakers 4L, 4R,
and provides the convolver 2L provided in the audio signal transmission path in order to correct
the characteristics of the speakers 4L, 4R to cancel and flatten the characteristics. By calculating
and controlling the 2R filter coefficients, the amplitude and phase are simultaneously corrected
to keep only a predetermined band of the transmission characteristic constant and to improve
the sound quality so as to reproduce the faithful original sound quality, for example, The
distortion of the sound image that can not be avoided by the speakers and headphones is
removed so that a natural audio signal can be enjoyed.
[0014]
Here, the filter coefficients of the convolvers 2L and 2R are calculated as coefficient data by the
measurement system shown in FIG.
That is, FIG. 2 is a measurement position corresponding to a listening position in the anechoic
chamber (not shown) in the state where the convolvers 2L and 2R are not provided by
connecting the switches 7L and 7R to the terminals ga and ha in FIG. The impulse response of
the speakers 4L and 4R at this position is measured by the microphone 8 provided, the response
characteristics of the speakers 4L and 4R are canceled, and the convolver 2L provided in the
audio signal transmission path is corrected to flat characteristics. FIG. 16 is a system
configuration diagram for realizing an ideal impulse response that simultaneously corrects the
amplitude and phase characteristics by calculating 2R filter coefficients and performing
convolution operation to correct response characteristics of the speakers 4L and 4R.
[0015]
In FIG. 2, 11 is a digital I / O board for transmitting ideal impulses as digital data, 5 is a DSP unit
for passing or convoluting the ideal impulses, 6 is an output of the DSP unit 5 / A converting D /
A converter, 7 is an amplifier for amplifying the converted signal and inputting to the speaker 4L
(or 4R), 8 is a microphone for taking in the signal output from the speaker 4L (or 4R), 9 is An
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amplifier for amplifying a signal taken in by the microphone 8, an A / D converter 10 for A / D
converting its amplified output, and an output from the A / D converter 10 through the digital I /
O board 11 and the computer 12 The speaker 4L is taken as an impulse response to the
workstation 13 and is not subjected to correction. Is carried out characteristic measurements
4R), the filter coefficients are calculated output as coefficient data of the measured impulse
response waveform based.
The characteristics of the microphone 8 are corrected in the process of calculation as necessary.
[0016]
That is, the I / O board 11 constitutes measurement signal generating means for generating a
measurement signal, and the DSP unit 5, the D / A converter 6, the amplifier 7, the speaker 4L,
and the paths of the microphone 8 to the work station 13 The configuration of the above
constitutes a response characteristic measuring means for obtaining the amplitude characteristic
and the phase characteristic which is the response of the audio signal transmission system based
on the measurement signal, and further, the workstation 13 flattens the amplitude of a
predetermined band among the obtained response characteristics. In addition, the computing
means is configured to determine the target characteristic replaced in the above and to obtain
the filter coefficient of the convolver provided in the audio signal transmission system so that the
response of the audio signal transmission system converges on the target characteristic. We have
realized a system that improves the sound quality by correcting it to the cancellation flat
characteristic.
[0017]
The impulse response of the loudspeakers 4L and 4R according to the configuration shown in
FIG. 2 is measured using the microphone 8 in the anechoic chamber, for example, using 4096
samples and performing synchronous addition 1000 times to suppress errors. Measured.
FIG. 3 shows an impulse response waveform h (t) obtained by the measurement system, and solid
lines shown in FIGS. 4 and 6 show amplitude characteristics and phase characteristics obtained
by Fourier transforming the impulse response waveform h (t).
[0018]
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Here, in the workstation 13 shown in FIG. 2, first, when the filter coefficient is determined, only
the mid frequency band of 200 Hz to 20000 Hz is flat with respect to the amplitude
characteristic ORI before correction shown by the solid line in FIG. A filter coefficient is
calculated by setting a target characteristic TAG indicated by a dotted line as a characteristic that
is substantially the same as the amplitude characteristic ORI for one or both of the low band and
the high band outside this band.
That is, an impulse response f0 (t) of the specific characteristic TAG indicated by a dotted line in
FIG. 4 which is an amplitude characteristic corrected based on the measurement characteristic
ORI (the amplitude characteristic related to the impulse response waveform h (t)) is determined.
From an impulse response f0 (t) and an expansion matrix H obtained from the impulse response
waveform h (t) and a matrix F0 having transposes HT and f0 (t) as one column, from one column
satisfying HTHG = HTF0 Let each element of the determinant G be the filter coefficients g (n) of
the convolvers 2L and 2R shown in FIG.
[0019]
The solution of the above determinant will be described below.
In this embodiment, the response waveform can be uniquely obtained on the time axis by finding
a solution that satisfies the determinant according to the above configuration.
Specifically, the least squares method using Levinson's algorithm (Reference: "Introduction to
application of digital filters", Journal of the Acoustical Society of Japan, vol. 43, No. 4 (1987),
Haruo Hamada) Filter coefficients that minimize the square of the difference between the
impulse responses to be obtained.
[0020]
Let g1, g2,..., Gm-1 be discrete coefficients of the impulse response of the convolver, the discrete
responses f0, f1,. It can be expressed.
[0021]
[Equation 1]
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[0022]
Where hi is the transfer characteristic, p is p = 0, 1,..., N + m-2.
Expressing equation (1) as a matrix,
[0023]
[Equation 2]
[0024]
Equation (2) can be further expressed as F = HG.
Here, taking the square of the difference between the impulse F0 of the input and the impulse
response F at the microphone position and taking an evaluation function P, P = (F−F0) T (F−F0)
= (HG−F0) T ( HG-F0) = (GTHT-F0T) (HG-F0) = GTHTHG-F0THG-GTHTF0 + F0TF0, and in order
to obtain the impulse response G of the convolver for minimizing the evaluation function P,
[0025]
[Equation 3]
[0026]
Calculate.
However, T represents that it is a transposition matrix.
Then, a solution G may be determined such that equation (4) = 0 to HTHG = HTF0 (5). That is, by
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8
setting the filter coefficient as in the above equation (5), the transmission characteristic is
corrected, the amplitude / phase characteristic at the microphone position becomes flat in the
desired band, and the low band and high band outside this band In the frequency band of the
same characteristic as the actual impulse response waveform is obtained. Thus, as shown by the
dotted lines in FIGS. 5 and 6, the amplitude characteristic and the phase characteristic provide
flat characteristics in the middle frequency band of 200 to 20000 Hz, and reproduce the faithful
original sound quality. The distortion of the sound image that can not be avoided by the speakers
and headphones can be removed, and in the low and high frequency bands other than the mid
band, the same characteristics as the actual speaker measurement system can be obtained, A
natural audio signal adapted in consideration of the characteristics of the speaker can be enjoyed
at the same time.
[0027]
Second Embodiment In the first embodiment described above, a target in which the amplitude of
only the predetermined middle frequency band of the measured amplitude characteristic is
replaced with flat is set, and the filter coefficient of the convolver is determined based on this. In
this second embodiment, the target characteristic of the first embodiment described above is
taken as the first target characteristic, while in the lower and higher bands than the
predetermined mid frequency band, the response is A second target characteristic in which the
amplitude of the characteristic is rolled off at a predetermined ratio is determined, and first and
second filter coefficients of the convolver corresponding to the first and second target
characteristics are calculated, The first and second amplitude characteristics corrected by
performing characteristic measurement based on the settings of the first and second filter
coefficients are determined, and the first and second amplitude characteristics are calculated.
The peak of the deviation value by searching each selects a filter coefficient having the smaller
the deviation by comparing the peak of their deviation value determined is determined.
[0028]
That is, in the lower and higher bands than the predetermined middle frequency band, the
amplitude characteristic is rolled off at a predetermined rate to enhance the low band and
attenuate the high band, so-called bus · · · By adding the treble characteristic and setting the
convolver's filter coefficient, it is possible to obtain a sound quality without distortion and obtain
a predetermined sound quality, and the effect of being able to correct the transmission path
while avoiding distortion mixing and lengthening of low and high frequencies. Can be expected,
but the characteristic measurement based on the setting of the filter coefficient in this case and
the setting of the filter coefficient according to the first embodiment selects the one with the
same coefficient length and the smaller deviation from the predetermined target. You will get one
that fits the characteristics of
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[0029]
FIG. 7 is a flow chart showing a control operation of selecting filter coefficients by the
workstation 13 based on the above comparison of deviations using the measurement system
shown in FIG.
First, the response of the speaker is measured by the measurement system shown in FIG. 2 (step
S61).
Based on the measurement results, the target characteristic in which the amplitude of only the
predetermined middle frequency band is replaced with flat similarly to the first embodiment is
used as the first target characteristic, and a band lower than the predetermined middle frequency
band And in the high band, the second target characteristics are set such that the response
characteristics are rolled off at a predetermined ratio (steps S62a and S62b), and the convolversa
corresponding to the first and second target characteristics are set. First and second filter
coefficients are respectively calculated (steps S63a and S63b).
[0030]
Next, the first and second amplitude characteristics are determined by respectively performing
characteristic measurement based on the first and second filter coefficients, that is, the
calculation is performed by inputting an impulse into the audio signal transmission system. The
first and second amplitude characteristics are determined by passing the convolver of the filter
coefficient (steps S64a and S64b), respectively, and the peak of the deviation value is searched
among the first and second amplitude characteristics (step S65a). , S65b), comparing the peaks
of the deviation values obtained (step S66), it is determined to select the filter coefficient with the
smaller deviation (steps S67, S68). By doing this, it is possible to select the one with the smaller
deviation from the predetermined target by the same coefficient length, and to obtain one that
conforms to the actual characteristic of the speaker.
[0031]
As described above, according to the audio signal transmission circuit of the present invention,
the audio signal transmission system including the speaker of the impulse response waveform h
(t) at the measurement microphone position is provided with a convolver and the measurement is
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performed. The impulse response f0 (t) of the specific characteristic at the microphone position
is set to be flat only in a predetermined middle frequency band and to have substantially the
same characteristics as the impulse response waveform h (t) outside the frequency band As a
result, flat characteristics can be obtained in the mid frequency band, faithful original sound
quality can be reproduced, distortion of the sound image that can not be avoided by the speakers
and headphones can be eliminated, and the mid frequency can be obtained. In the low and high
frequency bands other than the above, the same characteristics as the actual speaker
measurement system can be obtained, and adaptation is performed in consideration of the
characteristics of the actual speaker So it was there is an effect that it is possible to enjoy natural
audio signal at the same time.
[0032]
Further, according to the convolver coefficient computing device of the present invention,
measurement signal generating means for generating a measurement signal to be sent to an
audio signal transmission system, and an amplitude as a response of the audio signal
transmission system based on the measurement signal. Response characteristics measuring
means for obtaining characteristics and target characteristics in which the amplitude of only a
predetermined middle frequency band among the response characteristics obtained by the
response characteristics measuring means is replaced with flat are determined, and the response
of the audio signal transmission system And calculating means for determining the filter
coefficient of the convolver provided in the audio signal transmission system so as to converge
on the target characteristic, thereby realizing a system for improving the sound quality by
correcting the characteristic of the speaker to a flat characteristic. It is possible to relatively
easily obtain the convolver's filter coefficients suitable for use.
[0033]
Further, while the target characteristic in which the amplitude of only the predetermined middle
frequency band is replaced with the flat by the calculation means is regarded as the first target
characteristic, a band lower than the predetermined middle frequency band and In the high band,
the second target characteristic is determined by rolling off the response characteristic amplitude
at a predetermined rate, and the first and second filters of the convolver corresponding to the
first and second target characteristics. The first and second amplitude characteristics corrected
by respectively calculating the coefficients and performing the characteristic measurement based
on the setting of the first and second filter coefficients are determined, and the first and second
amplitude characteristics are calculated. Since the peaks of the deviation value among the above
are searched respectively, the determined peaks of the deviation values are compared, and it is
determined to select the filter coefficient with the smaller deviation. Flip select whichever
deviation is smaller for a given target by a factor length, it is possible to obtain the filter
coefficients adapted to the actual speaker-specific characteristics.
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[0034]
Brief description of the drawings
[0035]
1 is a block diagram showing the configuration of an embodiment of the audio signal
transmission circuit according to the first embodiment of the present invention.
[0036]
2 is a configuration diagram showing a measurement system of the filter coefficient of the
convolver according to the present invention.
[0037]
3 is an explanatory view showing an impulse response waveform before correction.
[0038]
4 is a characteristic diagram showing the amplitude characteristics and target characteristics
obtained by Fourier-transforming the impulse response waveform of FIG.
[0039]
5 is a characteristic diagram showing an amplitude characteristic obtained by Fouriertransforming the impulse response waveform of FIG. 3 and a corrected amplitude characteristic.
[0040]
6 is a characteristic diagram showing a phase characteristic obtained by Fourier-transforming the
impulse response waveform of FIG. 3 and a corrected phase characteristic.
[0041]
7 illustrates the selection operation of the filter coefficient as the second embodiment of the
present invention, the deviation based on the setting of the first and second target characteristics
by the workstation 13 using the measurement system shown in FIG. 2 It is a flowchart which
shows the control action which selects a filter factor by comparison.
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[0042]
Explanation of sign
[0043]
2L, 2R Convolver 4L, 4R Speaker 5 DSP unit (D / A converter 6, amplifier 7, speaker 4L,
microphone 8, amplifier 9, A / D converter 10, digital I / O board 11, computer 12, workstation
13 together with 6) D / A converter 7, 9 amplifier 7L, 7R switch 8 microphone 10 A / D
converter 11 digital I / O board (measurement signal generation means) 12 computer 13 work
station (calculation means) 15 control unit (constitutes coefficient supplying means with memory
16) 16 memory
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