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JPH0888893

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DESCRIPTION JPH0888893
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an
audio signal transmission circuit, and more particularly to an audio signal transmission circuit
which corrects the response characteristic of the speaker of a high fidelity audio apparatus and
reproduces a faithful original sound quality. .
[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]
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
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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. Although
this is intended to correct the response characteristics of the speaker, it aims to improve the
calculation when calculating the impulse response by inverse Fourier transform, and it is
intended to simply correct the frequency characteristics of the speaker flat. It is
[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
according to this disclosed technique, the amplitude characteristic and By performing
measurement and analysis so that the phase characteristics are both flat, for example, for
correction at high frequencies, the characteristics of the alias removal filter of the D / A
converter used in the measurement system are included in the measurement data The
characteristics of this high-frequency part are sharply raised, and if you try to listen to the ones
that are actually corrected, the distortion correction is not clearly felt, and the sound quality is
rather distorted.
[0006]
Further, according to the technique disclosed in Japanese Patent Application Laid-Open No. 63281510, since the frequency is kept flat in the low frequency region up to the operating limit of
the speaker, the distortion is heard in the low frequency region.
[0007]
Furthermore, according to the conventional example of Japanese Patent Laid-Open No. 2272819, although the correction limit of each band in the band can be detected to minimize the
correction error, the speaker in the low frequency region peculiar to the speaker can be
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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 operating limit.
[0008]
The present invention has been made to solve the problems relating to the above-described
conventional example, and distortion of the sound image which can not be avoided by the
speaker or the headphone is removed to simultaneously correct the amplitude and the phase, and
the transmission characteristics are determined. It is an object of the present invention to obtain
an audio signal transmission circuit which can enjoy natural audio signals by maintaining only a
middle frequency band constant and reproducing a faithful source sound quality.
[0009]
SUMMARY OF THE INVENTION In order to achieve the above object, an audio signal
transmission circuit according to the present invention comprises an audio signal transmission
system including a speaker of an impulse response waveform h (t) at a measurement microphone
position. A convolver is provided, and an expanding matrix H obtained from impulse responses f0
(t) and h (t) of a specific characteristic at the measurement microphone position, its transposed
matrix HT and a matrix F0 having the impulse response f0 (t) as one column In an audio signal
transmission circuit in which each element of a matrix G consisting of one column satisfying =
HTF0 is set to the filter coefficient g (n) of the convolver, the impulse response f0 (t) of the
specific characteristic depends on the speaker characteristic By selecting only a predetermined
mid frequency band to be flat, the transmission characteristic is kept constant only for a
predetermined mid frequency band. And is characterized.
[0010]
Further, the filter coefficient of the convolver is set so that the amplitude characteristic is rolled
off at a predetermined rate in a band lower than a predetermined middle frequency band.
[0011]
Further, the filter coefficient of the convolver is set so that the amplitude characteristic is rolled
off at a predetermined rate in a band higher than a predetermined middle frequency band.
[0012]
DESCRIPTION OF THE PREFERRED EMBODIMENTS First Embodiment FIG. 1 is a block diagram
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showing a first embodiment of an audio signal transmission circuit according to the present
invention.
In FIG. 1, two-channel stereo sound sources 1L and 1R generate desired audio signals.
Convolvers 2L and 2R for respectively correcting the output characteristics of the speakers 4L
and 4R to simultaneously correct the amplitude and phase characteristics are connected to the
speakers 4L and 4R via the amplifiers 3L and 3R.
5 performs switching control of a switch to be described later and measures the output
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 for correcting the response
characteristic of the speaker.
A storage unit (memory) 6 is connected to the control unit 5 to store correction filter coefficients
obtained based on the measurement of the output characteristics of the speakers 4L and 4R.
7L and 7R separate the convolvers 2L and 2R from the transmission line of the audio signal
when measuring the output characteristics of the speakers 4L and 4R based on the control of the
control unit 5, and transmit the audio line of the audio signal when correcting the output
characteristic of the speaker Is a switch for switching so as to provide the convolvers 2L and 2R.
[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]
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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 measures the impulse response of the speakers 4L and 4R at the microphone
position in a state where the switches 7L and 7R are connected to the terminals ga and ha
respectively and the convolvers 2L and 2R are not provided in FIG. In order to correct the
characteristics of the speakers 4L and 4R to a flat characteristic, the filter coefficients of the
convolvers 2L and 2R provided in the transmission path of the audio signal are calculated and
convolution operation is performed to correct the speaker characteristics to thereby correct the
amplitude and phase characteristics. Is a system configuration diagram for realizing an ideal
impulse response that simultaneously corrects
[0015]
In FIG. 2, 11 is a digital I / O board for transmitting ideal impulses as digital data, 5 is a DSP unit
for ideal impulse passing (through path) or convolver processing, 6 D / A converts its output. 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 a
microphone 8 An amplifier for amplifying the taken signal, 10 is an A / D converter for A / D
converting the amplified output, and the output from the A / D converter 10 is through the
digital I / O board 11 and the computer 12 The characteristic measurement of the speaker 4L (or
4R) taken into the workstation 13 as an impulse response and before being corrected is We, the
filter coefficients are calculated output as coefficient data based on the measured impulse
response waveform.
The characteristics of the microphone 8 are corrected in the process of calculation as necessary.
[0016]
The impulse response of the loudspeakers 4L and 4R according to the configuration shown in
FIG. 2 described above is measured using a microphone in an anechoic chamber, for example,
using 4096 samples and performing synchronous addition 1000 times to make an error. It
measures and holds down.
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FIG. 3 shows the impulse response waveform h (t) obtained by the measurement system as
described above, and the solid lines shown in FIG. 4 and FIG. 5 show amplitude characteristics
and phase characteristics obtained by Fourier transforming the impulse response waveform h (t).
.
[0017]
Here, the workstation 13 shown in FIG. 2 has the amplitude before the correction shown in FIG. 3
when obtaining the filter coefficient in order to obtain the inverse characteristic INV with respect
to the middle frequency band of 200 Hz to 20000 Hz shown in FIG. The impulse of the specific
characteristic TAG which is the final target characteristic shown in FIG. 6 corrected with the
target characteristic with the inverse characteristic INV shown in FIG. 5 as the target
characteristic with respect to the characteristic ORI (amplitude characteristic relating to the
impulse response waveform h (t)) A response f0 (t) is obtained, and an expansion matrix H
obtained from the impulse response f0 (t) of the specific characteristic and the impulse response
waveform h (t), and a matrix in which transposed matrices HT and f0 (t) thereof are in one
column Let F0 be the filter coefficient g (n) of the convolvers 2L and 2R shown in FIG.
For less than 200 Hz outside the band, the cutoff frequency is not flat but 50 Hz, the reverse rolloff characteristic of approximately 30 dB / OCT, and the rapid roll-off characteristic at 20 kHz or
more.
[0018]
The solution of the above determinant will be described below. In the present embodiment, the
response waveform is uniquely obtained on the time axis by finding a solution that satisfies the
determinant according to the above-described configuration. Specifically, according to Levinson's
least squares method (Reference: "Introduction to application of digital filters", Journal of the
Acoustical Society of Japan, vol. 43, No. 4 (1987), Haruo Hamada), impulse responses obtained at
input and output ends of convolvers Filter coefficients that minimize the square of the difference
of
[0019]
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Now, let g1, g2, ..., gm-1 be discrete coefficients of the impulse response of the convolver, the
discrete responses f0, f1, ... fn + m-2 at the microphone position are It can be expressed.
[0020]
[Equation 1]
[0021]
Where hi is the transfer characteristic, p is p = 0, 1,..., N + m-2.
Expressing equation (1) as a matrix,
[0022]
[Equation 2]
[0023]
Equation (2) can be further expressed as F = HG.
Here, taking the square of the difference between the target impulse F0 and the impulse
response F calculated from the obtained coefficient, and assuming 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,
[0024]
[Equation 3]
[0025]
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Where T represents a transposed matrix.
Calculate
Then, a solution G may be determined such that equation (4) = 0 to HTHG = HTF0 (5). That is, by
setting the filter coefficient as in the above equation (5), the transmission characteristic is
corrected, and the amplitude / phase characteristic at the microphone position becomes flat in
the desired band. As described above, since the rapid roll-off characteristic is selected for 20 kHz
or more, the characteristic of the alias removal filter of the D / A converter used in the
measurement system can be prevented from being included in the measurement data. It does not
have the characteristic of rapidly lifting up, and if you try to listen to the one that is actually
corrected, the sound quality will not be distorted. Further, in the low frequency region, since it is
not intended to keep the frequency flat below the operation limit of the speaker, the distortion in
the low region is not increased and can not be heard.
[0026]
FIGS. 8 to 10 show an impulse response waveform whose transmission characteristic has been
corrected based on the setting of the filter coefficient as described above, and an amplitude
characteristic and a phase characteristic obtained by Fourier transforming the impulse response
waveform. As shown in FIG. 10, in the amplitude characteristic and the phase characteristic, flat
characteristics are obtained in the middle frequency band of 200 to 20000 Hz, and the faithful
original sound quality is reproduced, and can not be avoided in the speaker or headphone The
distortion of the sound image can be removed to enjoy a natural audio signal simultaneously.
[0027]
Second Embodiment Next, in the second embodiment, the specific characteristic which is the final
target characteristic in the first embodiment is selected as indicated by TAG2 shown by a dotted
line in FIG.
In this way, raising the low range, gently dropping the high range from about 16 kHz and rolling
off sharply at 20 kHz or more has the advantage of providing a richly reproduced sound in
addition to the advantages of the first embodiment. . In addition, it is possible to realize without
increasing the scale of the coefficient length of the convolver. Note that the roll-off
characteristics in the low frequency range can be obtained with reference to the roll-off
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characteristics specific to the speaker used.
[0028]
That is, in the second embodiment, the selection of the reference sets the filter coefficient of the
convolver so that the amplitude characteristic rolls off at a predetermined rate in a band lower
and higher than a predetermined middle frequency band. By setting the convolver's filter
coefficients so as to enhance the low range and attenuate the high range and add so-called bass
treble characteristics, it is possible to obtain distortion-free tone and to obtain predetermined
tone. There is an effect that correction of the transmission path can be performed while avoiding
distortion and increase in distortion in the low band and high band.
[0029]
As described above, according to the present invention, the impulse response f0 (t) of the specific
characteristic is dependent on the speaker characteristic so that only a predetermined mid
frequency band is flat. Reference h '(selected with an inverse characteristic as a target
characteristic for the impulse response waveform h (t) at the microphone position and corrected
by the impulse response f0 (t) of the specific characteristic which is the final target characteristic
corrected with this target characteristic t) to obtain an impulse response f0 (t) of the specific
characteristic, and to obtain an expansion matrix H from f0 (t) and the impulse response
waveform h (t), the expansion matrix H and its transposed matrix HT And each element of the
matrix G consisting of one column satisfying HTHG = HTF0 is set as the filter coefficient of the
convolver by the matrix F0 having the impulse response f0 (t) as one column Therefore,
distortions in the sound image that can not be avoided by the speakers and headphones are
removed to simultaneously correct the amplitude and phase, and only the predetermined middle
frequency band of the transmission characteristics is kept constant to reproduce the faithful
original sound quality. This has the effect of being able to enjoy natural audio signals.
Also, by setting the convolver's filter coefficient so that the amplitude characteristic rolls off at a
predetermined rate in the lower and higher bands than the predetermined mid-range frequency
band, low- and high-range distortion mixing and lengthening Correction of the transmission line
while avoiding the In addition, there is an effect that bus / treble characteristics can be
simultaneously added by setting the characteristics.
[0030]
Brief description of the drawings
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[0031]
1 is a block diagram showing an audio signal transmission circuit according to the present
invention.
[0032]
2 is a configuration diagram showing a measurement system of the filter coefficient of the
convolver according to the present invention.
[0033]
3 is an explanatory view showing an impulse response waveform before correction.
[0034]
4 is a characteristic diagram showing the amplitude characteristic obtained by Fourier transform
of the impulse response waveform of FIG.
[0035]
5 is a characteristic diagram showing the inverse characteristic.
[0036]
6 is a characteristic diagram showing the final target characteristics.
[0037]
7 is a characteristic diagram showing the phase characteristics of the impulse response
waveform of FIG. 3 after Fourier transform.
[0038]
8 is an explanatory view showing an impulse response waveform according to the correction
result according to the present invention.
[0039]
9 is a characteristic diagram showing the amplitude characteristic obtained by Fourier transform
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of the impulse response waveform of FIG.
[0040]
10 is a characteristic diagram showing the phase characteristics of the impulse response
waveform of FIG. 8 after Fourier transform.
[0041]
Explanation of sign
[0042]
2L, 2R Convolver 4L, 4R Speaker 5 Control unit 7L, 7R Switcher 8 Microphone 13 Workstation.
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