close

Вход

Забыли?

вход по аккаунту

JPH1051890

код для вставкиСкачать
Patent Translate
Powered by EPO and Google
Notice
This translation is machine-generated. It cannot be guaranteed that it is intelligible, accurate,
complete, reliable or fit for specific purposes. Critical decisions, such as commercially relevant or
financial decisions, should not be based on machine-translation output.
DESCRIPTION JPH1051890
[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
that corrects the response characteristics of the speaker of a high fidelity audio device and
reproduces faithful source sound quality. The present invention relates to a convolver coefficient
calculation device for obtaining convolver coefficients.
[0002]
2. Description of the Related Art Conventionally, by using a digital filter in a speaker system, the
sound pressure and group delay characteristics of the speaker system are flattened to correct the
response characteristics of the speaker and to reproduce the faithful original sound quality.
Devices have been implemented.
[0003]
By the way, when the sound pressure and group delay characteristics of the speaker system are
flattened, the following three main factors can be mentioned as the factors that impede the
flattening.
1) Factors due to diffraction and reflection of the speaker cabinet. 2) Factors caused by phase
rotation by the analog filter. 3) A factor due to the divided vibration of the diaphragm (including
08-05-2019
1
the surrounding support system) in the speaker unit. Among these factors, the factors 1) and 2)
can be corrected without any problem.
[0004]
However, the factor 3) is a fairly difficult correction. The divided vibration that causes this 3)
tends to occur especially in the high frequency band, and in the case of such high frequency, the
divided vibration is generated according to each unique vibration mode of the speaker cone, and
so on. The complex vibration of It is difficult to suppress such complicated vibrations simply by
averaging the sound pressure levels. Therefore, there is a problem that the viewer can not obtain
high fidelity sound source reproduction on the high frequency side. Therefore, the present
invention is to provide an audio signal transmission circuit that solves such problems.
[0005]
SUMMARY OF THE INVENTION The present invention is made based on the following basic idea.
Although it is ideal to use a band in which the diaphragm vibrates as an integral part of the
speaker, in practice it is often the case that a band vibrating with a peak and a dip must also be
used. The above factors 1) and 2) are considered to be a zone where division vibration hardly
occurs and the entire cone vibrates, and it can be considered that the problem can be solved by
averaging peaks and dips.
[0006]
Next, since the divided vibration mentioned in the above 3) occurs in the above-mentioned band
which causes a problem, the speaker cone vibrates as a whole, and moreover, it is assumed that a
surge is caused due to each natural vibration. It is conceivable that simply averaging the peaks
and dips as in the solutions of the above 1) and 2) may lead to a complicated undulation.
Therefore, in this band, the concept is to prevent the generation of complex waves and keep the
sound pressure substantially flat by suppressing only the sound pressure above the average level
and leaving the sound pressure below the average level as it is. It is.
[0007]
As the solution means, an audio signal from a predetermined sound source is given to the
08-05-2019
2
transducer using a convolver that convolves and outputs a signal from a sound source according
to a set coefficient, and a transducer. A predetermined characteristic of the impulse response f0
(t) of the impulse response waveform h (t) at the measurement position based on the impulse
response waveform h (t) measured in advance at the measurement position near the transducer
In the middle band, the lower band than the predetermined band becomes flat, and only the level
higher than the predetermined level in the higher band is suppressed to the predetermined level,
and the calculated inverse filter coefficient is supplied to the convolver And providing an audio
signal transmission circuit.
[0008]
DESCRIPTION OF THE PREFERRED EMBODIMENT The preferred embodiment of the present
invention will be described by way of a preferred embodiment.
FIG. 1 is a block diagram showing an embodiment of the audio signal transmission circuit of the
present invention. In the figure, sound sources 1L and 1R of the two-channel stereo system 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.
[0009]
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 applies the calculated 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.
[0010]
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. Switches 7L and 7R are provided to separate the convolvers 2L
and 2R from the transmission line of the above and to provide the convolvers 2L and 2R in the
transmission line of the audio signal when correcting the response characteristics of the speakers
4L and 4R. .
08-05-2019
3
[0011]
That is, in the configuration shown in FIG. 1, the impulse response of the speakers 4L and 4R is
measured, and the convolvers 2L and 2R provided in the transmission path of the audio signal in
order to correct the characteristics of the speakers 4L and 4R to cancel and flatten the
characteristics. By calculating and controlling the filter coefficient, the amplitude and the 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, a speaker or headphone It is intended to remove the distortion of the
sound image that can not be avoided in order to be able to enjoy a natural audio signal.
[0012]
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. , 2R filter coefficients are
calculated, and a convolution operation is performed to correct response characteristics of the
speakers 4L and 4R, thereby achieving an ideal impulse response for correcting the amplitude
and phase characteristics.
[0013]
In FIG. 2, 11 is a digital I / O board that sends ideal impulses as digital data, 6 is a D / A
converter that D / A converts the output of digital I / O board 11, and 7 is the converted signal
Amplifier for amplification and input 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 amplifier for amplifying the signal taken in
for the microphone 8, 10 is its amplified output In the A / D converter for A / D conversion, the
output from the A / D converter 10 is taken as an impulse response to the workstation 13
through the digital I / O board 11 and the computer 12 and is a speaker before correction The
characteristic measurement of 4L (or 4R) is performed, and the filter coefficient is converted to
08-05-2019
4
coefficient data based on the measured impulse response waveform. It is calculated output as.
The characteristics of the microphone 8 are corrected in the process of calculation as necessary.
[0014]
That is, the I / O board 11 constitutes measurement signal generating means for generating a
measurement signal, and the configurations of the D / A converter 6, the amplifier 7, the speaker
4L, and the microphones 8 to 13 are based on the measurement signals. The target is a response
characteristic measuring means for obtaining the amplitude characteristic and the phase
characteristic which is the response of the audio signal transmission system, and the workstation
13 is replaced so as to average the amplitude of a predetermined band among the obtained
response characteristics. The arithmetic means is configured to determine the characteristics 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, substantially canceling the speaker characteristics. We have realized a system that
improves the sound quality by correcting to the flattened characteristics.
[0015]
The impulse responses of the speakers 4L and 4R according to the configuration shown in FIG. 2
are measured using the microphone 8 in the anechoic chamber, for example, using 4096 samples
and performing synchronous addition 1000 times to suppress errors. Ru.
FIG. 3 shows an impulse response waveform h (t) obtained by such a measurement system, and
FIG. 4 (a) is a waveform showing an amplitude characteristic obtained by Fourier transforming
the impulse response waveform h (t). Shows a waveform of an impulse response f0 (t) of a
specific characteristic to be a target.
[0016]
Here, in the work station 13 shown in FIG. 2, when the filter coefficient is first determined, the
mid band of about 150 Hz to about 8000 should be substantially averaged with respect to the
amplitude characteristic before the correction shown in FIG. The low band and high band other
than this band are considered as uncorrectable bands.
08-05-2019
5
[0017]
And the so-called piston band (sound pressure without a wave of the speaker cone) caused by 1)
and 2) which pointed out the frequency band of about 150 Hz to about 1600 Hz in the section of
the above-mentioned problems among the above correction bands According to the above, the
level of the band that vibrates as a whole is averaged, and the band beyond this is regarded as
the band where the speaker The filter coefficient is calculated by setting the target level as a
target characteristic that suppresses the level to an average level.
[0018]
That is, the impulse response f0 (t) of the specific characteristic shown in FIG. 4 (b) which is the
amplitude characteristic corrected based on the measurement characteristic (the amplitude
characteristic according to the impulse response waveform h (t)) is determined. One column
satisfying HT HG = HT F0 by an impulse response f0 (t) and the expansion matrix H obtained
from the impulse response waveform h (t) and a matrix F0 in which the transposed matrices HT
and f0 (t) are one column Each element of the determinant G consisting of the filter coefficients g
(n) of the convolvers 2L and 2R shown in FIG.
[0019]
The solution of the above determinant is 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]
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 can be expressed by the
following equations.
08-05-2019
6
[0021]
Where hi is the transfer characteristic, p is p = 0, 1,..., N + m-2.
Expressing equation (1) as a matrix,
[0022]
Equation (2) can be further expressed as:
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) = GTHT HG-F0THG-GTHT F0 + F0TF0, and in order
to obtain the impulse response G of the convolver for which the evaluation function P is
minimized,
[0023]
Calculate.
However, T represents that it is a transposition matrix. Then, a solution G may be determined
such that HT HG = HT F0 (5) from equation (4) = 0. That is, by setting the filter coefficient as in
the above equation (5), the transmission characteristic is corrected, the amplitude / phase
characteristic at the microphone position is averaged in the piston band, and a level higher than
the average level in the higher band Correction is performed so as to suppress to the average
level, and the low band and high band other than these are not corrected as uncorrectable bands,
and have the same characteristics as an actual impulse response waveform. In this way, faithful
original sound quality can be reproduced, 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 actual speakers etc. The same characteristics as the measurement system
are obtained, and natural audio signals adapted in consideration of the characteristics of the
actual speaker can be enjoyed simultaneously.
08-05-2019
7
[0024]
FIG. 5 is a flow chart showing a control operation of setting the filter coefficient by the
workstation 13 using the measurement system shown in FIG. First, the response of the speaker is
measured by the measurement system shown in FIG. 2 (step S1), and this impulse response
waveform h (t) is Fourier transformed (FFT: frequency-time axis transformation) to obtain an
amplitude characteristic (step S2). ). Next, in step S3, the low band and high band not to be
corrected are determined, and the level of each point in these bands is determined.
[0025]
On the other hand, in step S4, the average level of the piston zone is determined. Here, averaging
of all points may be performed, or averaging of arbitrary intervals may be performed. Next, this
averaging level, for example, 88 dB, is set as the piston band level (step S5).
[0026]
Then, a band for which ka> 1 is estimated to be generated is set. That is, the frequency from the
frequency sufficiently higher than the low band resonance (in general, the inertial control region)
to the frequency at which ka = 1 is set.
[0027]
Next, in step S7, the conversion result after measurement obtained in step S2 is compared with
the average level, and if the conversion result level is smaller, the process proceeds directly to
step S9, and the conversion result level is high. If there is, the process proceeds to step S8, and
the level of each point, that is, the above-mentioned average level is set.
[0028]
In step S9, low- and high-frequency point data outside the correction band and the correction
data obtained in step S8 are synthesized, and the synthesized data is subjected to inverse
frequency transform (IFFT) (steps S9, 10) ).
08-05-2019
8
Finally, in step S10, the target frequency / measurement frequency is calculated by the method
of least squares to obtain a convolver coefficient.
[0029]
As described above, in this embodiment, the sound source in the high-pitched range can be
reproduced faithfully by performing speech processing on the basis of the coefficients that can
obtain the characteristics as described above. In the above embodiment, the sound pressure
higher than the average level is corrected to the average level in the band exceeding the piston
band, but the sound pressure is not limited to the average level and is corrected to conform to
the preset envelope characteristic. It is good. In the above embodiment, the measurement system
is provided in the audio signal transmission circuit system to determine the coefficient, but the
measurement system is separately and independently prepared, and the coefficient is determined
in advance by the method described above, The coefficients may be stored in the memory 16.
[0030]
As described above, according to the audio signal transmission circuit of the present invention, in
particular, a predetermined level or more of the sound pressure in the high band where the
waviness occurs in the speaker cone is suppressed to the predetermined level. So, you can
faithfully reproduce high-range sound sources.
[0031]
Brief description of the drawings
[0032]
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.
[0033]
2 is a configuration diagram showing a measurement system of the filter coefficient of the
convolver according to the present invention.
08-05-2019
9
[0034]
3 is an explanatory view showing an impulse response waveform before correction.
[0035]
4 is a characteristic diagram showing the amplitude characteristics and target characteristics
obtained by Fourier-transforming the impulse response waveform of FIG.
[0036]
5 is a flowchart for calculating the filter coefficient by the workstation 13 using the measurement
system shown in FIG.
[0037]
Explanation of sign
[0038]
2L, 2R Convolver 4L, 4R Speaker 7, 9 Amplifier 7L, 7R Switcher 8 Microphone 10 A / D
Converter 11 Digital I / O Board (Measurement Signal Generation Means) 12 Computer 13
Workstation (Calculation Means) 15 Control Unit (Memory) 16) (coefficient supply means
together with 16)
08-05-2019
10
1/--страниц
Пожаловаться на содержимое документа