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JPS63300699

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DESCRIPTION JPS63300699
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
network for a multi-way speaker device in which the design of a filter is facilitated and the
amount of hardware is reduced by improving a dividing network circuit for driving a speaker
with digital signals. PRIOR ART A conventional dividing network circuit is shown in FIG. In the
figure, reference numeral 1 denotes an analog input terminal to which an analog signal from a
tuner, an analog tape recorder, an analog player or the like is inputted, and 2 denotes an analog /
digital converter (hereinafter referred to as A / 3) is a digital audio tape recorder, 3 is a digital
source source having a digital output such as a compact disc, 4 is a switch for selectively
switching between the A / D converter 2 and the digital source source 3, 5 is a high pass filter 6
7 and a master clock oscillator for performing arithmetic processing of the low pass filters 8 and
9. When the sampling clock frequency of the digital signal group selectively input by the switch
4 is different for each input source by the switch 4, 22 Number for converting to one certain
clock frequency It is a ring converter. If the clock frequency of the output from the switch 4 is
the same, the sampling converter 22 is unnecessary. 6 and 7 are high-pass filter circuits for
performing arithmetic processing to give an arbitrary attenuation characteristic on the low
frequency side on the frequency axis with respect to an input digital signal group of sampling
clock frequency (fs); A low pass filter circuit that performs arithmetic processing to give an
arbitrary attenuation characteristic on the high frequency side of the input digital signal group of
clock frequency (fs) on the frequency axis, and 10 to 13 are the above-mentioned high pass filter
circuits Digital / analog converters (hereinafter referred to as D / A converters) for converting
digital signals from the 6 ° 7 and low pass filter circuits 8 and 9 into analog signals respectively,
and 14 to 17 from the D / A converters 10 to 13 An amplifier for amplifying an analog signal, 18
° 19 is left and right high-range speakers driven by the output from the amplifier 14.16, 20.21
is the above-mentioned increase A left and right low-frequency speaker that is driven by the
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output of more vessels 15.17. Thus, in the conventional dividing network of the above
configuration, the high-pass filter 6.7 extracts only the signals of the left and right high band
components of the digital signal selected by the switch 4, and low Only the low pass component
signals on the left and right are extracted by the low pass filters 8 and 9.
Then, the D / A converters 10 to 13 convert the high frequency left and right analog signals and
the low frequency left and right analog signals, and the next stage amplifiers 14 to 17 amplify
the signals to drive the speakers 8 to 21. [Problems to be Solved by the Invention] By the way, in
the above-mentioned conventional de-hinging network described above, various configurations
can be considered as a configuration for obtaining desired amplitude and phase characteristics as
the high pass filter 6.7. When only non-recursive filters are used, it is necessary to increase the
order of the filters to increase the circuit scale. Therefore, there is a problem that it is necessary
to design so as to increase the processing speed of calculation, which makes the design difficult.
In addition, even when only a cyclic filter is used, it is necessary to increase the order and the
number of stages of the filter, and there is a problem that the circuit scale is increased and the
design is not easy. SUMMARY OF THE INVENTION The present invention is intended to solve the
above-mentioned problems, and by combining a non-recursive filter and a recursive filter, the
circuit scale as a filter can be reduced and the desired amplitude, It is an object of the present
invention to provide a network for a multi-way speaker device that can obtain phase
characteristics and can easily design a filter. SUMMARY OF THE INVENTION In order to achieve
the above object, according to the present invention, a filter into which a digital signal is input is
a cyclic filter having desired phase characteristics for correcting the phase characteristics of a
speaker unit, and the cyclic type. The gist of the present invention is to correct the amplitude
characteristic of the filter and to construct a linear phase characteristic non-recursive filter
having a desired amplitude characteristic. DETAILED DESCRIPTION OF THE PREFERRED
EMBODIMENT An embodiment of the present invention will be described below with reference to
FIGS. The same reference numerals as in FIG. 8 indicate the same parts and the explanation will
be omitted. The difference between the present invention and the prior art is that the sampling
converter 23 for converting to a clock frequency lower than the clock frequency (fs) of the digital
signal input via the switch 4 is connected to the front stage of the low pass filter 8 An A / D
converter 24.25 for converting a digital signal to an analog signal with the low sampling signal is
connected to a subsequent stage of the low pass filter 8, and a high pass filter 6.7 is further
shown in FIG. It is composed of cyclic filters 6a and 7a for correcting the phase characteristic of
the speaker unit, and non-recursive filters 6b and 7b for correcting the amplitude characteristics
of the cyclic filters 6a and 7a.
Reference numerals 26 to 29 denote delay circuits connected to the front stages of the respective
D / A converters 10 ° 11.24.25. Next, to explain the operation based on the above
configuration, the digital signal selected by the switch 4 is applied to the left and right high pass
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filters 6.7 as in the above-mentioned conventional example. Here, as shown in FIG. 2, the high
pass filter 6.7 is composed of non-recursive filters 6b and 7b and cyclic filters 6a and 7a, so that
the phase of the speaker unit as shown in FIG. In order to correct the characteristics (indicated by
the broken line, and the solid line indicates the amplitude characteristic), the phase
characteristics as shown in FIG. 5a are realized by the recursive filters 6a and 7a. Since these
cyclic filters 6a and 7a have the characteristics of the ropes-to-list because they have the
amplitude characteristics as shown in FIG. 5 (bl, but there is no need for such an amplitude
characteristic, these amplitude characteristics are corrected You will need to Therefore, the nonrecursive filters 6b and 7b are designed so as to correct the amplitude characteristic and to
obtain the desired characteristic of the high pass filter 6.7 as a whole. The characteristics of the
non-recursive filters 6b and 7b are shown in FIG. That is, the characteristic of (.delta.) And the
characteristic of (d) in FIG. 6 are mixed to obtain a characteristic such as C1, and the amplitude
characteristic is corrected by the non-recursive filters 6b and 7b having this characteristic. It is.
Since the non-recursive filters 6b and 7b have linear phase characteristics, the non-recursive
filters 6b and 7b do not affect the phase characteristics of the recursive filters 5a and 7a.
Therefore, since the desired phase characteristics are obtained by the recursive filters 6a and 7a
and the desired amplitude characteristics are obtained by the non-recursive filters 6b and 7b, the
desired phase and amplitude characteristics as shown in FIG. The high pass filter 6.7 can be
produced. On the other hand, the signal is converted to a sampling frequency lower than the
clock frequency (fs) of the digital signal input by the sampling converter 23 and added to the low
pass filter 8. Here, by lowering the clock frequency to 1 / n by the sampling converter 23, the
time that can be used for signal processing becomes n times compared to that before the clock
frequency is lowered. In the present embodiment, the clock frequency of the input digital signal
group is fs, and the clock frequency is lowered to fs / 2 by the sampling converter 23. Thus,
hardware of the same size as the conventional example is used. The low pass filter 8 requires
only half the time required for signal processing.
In addition, if the low pass filter 8 is used in time division processing, filtering processing of the
left and right channels can be performed. As a result, the low pass filter 9 becomes unnecessary
as in the prior art. However, in this case, the shift register of the order bone (number of filter
stages) of the low pass filter 8 is only doubled, and the cost does not change so much. Low-pass
filters generally used in multi-way speaker systems have a sufficiently low cutoff frequency to the
clock frequency (f-44, 1 KHz) ratio of the input digital signal group, so 1 / n (n is 2 The above
integers) and n are relatively high even at 4, 8 and the design of the cut-off frequency
characteristic becomes easy by lowering the sampling clock frequency low. For example, digital
signal processing group input by the sampling converter 23 For example, when fs is 172 and fs /
2 is used to reduce the clock frequency of f, the frequency from fs / 2 to fs is not sufficiently
attenuated, and the folding frequency is within the use band of 0-fs / 2. It will cause distortion.
Therefore, in the case of reducing to 1⁄2, after attenuating frequency components of fs / 2 or
more, every other data is sampled from the data of the sample string. Thus, the low frequency
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digital signals of the left and right channels extracted by the low pass filter 8 are applied to the
digital delay circuits 28 and 29. Here, the delay circuits 28 and 29 are provided because the
difference in arrival time occurs at the listening position due to the difference between the sound
source positions of the bass speaker 20 or 21 and the treble speaker 18 or 19, and waveform
distortion occurs. It is to prevent. That is, as shown in FIG. 3, when the high-pitched speakers 18
and 19 are behind the low-pitched speaker 20.21 as viewed from the sound receiving point P,
the low-pitched speaker 20.21 and the high-pitched speaker 18.19 are used. Assuming that
T.beta. And T.sub.h are T.sub..beta. And T.sub.h, respectively, TN <Th, and delay time difference
.DELTA.S occurs in the sound from the high-pitched speaker 18.19 particularly near the
crossover, that is, .DELTA.5 = Th. Cause waveform distortion. A delay circuit 28 ° 29 is provided
to correct this time difference ΔS. In addition, the delay circuit 26.27 passes the digital signal
input through the switch 4 to the high pass filter 6.degree. 7 and the low pass filter 8. At this
time, a difference occurs in the signal processing time required for the filtering. Provided to
correct the That is, the time required to process a digital signal by high-pass filter 6.7 is the same
as TH1. The time required to process low-pass filter 8 is TL, and the time required to
undersample by sampling converter 23 It is TU.
Here, if TU + TL> TH, and the difference Δf in the processing time in this case, then Δf− (TU +
TL) −TH, and if Δf> ΔS, then the bass speaker 20.21 and the high band speakers 18 and 19 On
the other hand, at the sound receiving point P, there is a delay time difference in the sound from
the bass speaker 20.21. Therefore, in such a case, a delay circuit 26.27 is provided downstream
of the high pass filter 6.degree. 7 to correct this time difference .DELTA.f. In this case, the delay
circuits 28 and 29 are unnecessary. In the case of Δf−ΔS, the delay circuits 26 to 29 are not
necessary in any system of the high pass filter 6 ° 7 and the low pass filter 8. The digital signals
subjected to time axis correction by the delay circuits 26 to 29 are converted to analog signals by
the D / A converters 1 ° and 11 and 24.25, respectively, amplified by the amplifiers 14 to 17,
and amplified by the speakers 18 to Added to 21. Different filters are used in D / A converters
10.11 and 24.25 which convert this digital signal into an analog signal. That is, since the digital
signal output from the low pass filter 8 has a sampling clock frequency lower than the sampling
frequency sampled and input by the sampling converter 23, the D / A converter 24.25 is a D / A
converter 10 Not the same as .11. The signals converted into analog signals are amplified by the
amplifiers 14 to 17 to drive the speakers 18 to 21. As another embodiment, by connecting the
sampling converter after the low pass filter 8 to return to the original sampling clock frequency,
the D / A converters 10, 11, 24 and 25 can use the same clock frequency. . Also, the multi-way
speaker system described so far is a two-way speaker system for bass and treble, but in the case
of a multi-way speaker system in which dedicated speakers are provided separately for each used
band, a band pass filter (BPF matched to the used band) ), And the delay circuit may be provided
after the BPF circuit to provide a network for a multi-way speaker device in which the time axis
at the sound receiving point is aligned. In the above-described embodiment, although the filter in
which the recursive filter and the non-recursive filter are combined is used only for the high pass
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filter, it is also applicable to the low pass filter. Also, from the input side, the recursive filter 6a.
7a, non-recursive filters 6b, and 7b may be arranged in this order.
As described above, the present invention corrects the phase characteristics of the speaker unit
in the dividing network in which high- and low-pass signals are separated by the high-pass filter
and the low-pass iI filter. By comprising a recursive filter and a non-recursive filter that corrects
the amplitude characteristics of the cyclic filter, desired phase and amplitude characteristics can
be realized with a small circuit scale, and filter design can be easily performed. And other effects.
[0002]
Brief description of the drawings
[0003]
FIG. 1 is a block diagram showing an embodiment of a network for a multi-way speaker device
according to the present invention, FIG. 2 is a detailed block diagram of a part of the above, and
FIG. FIG. 4 is a characteristic diagram of the speaker system, FIG. 5 is a characteristic diagram of
the recursive filter, FIG. 6 is a characteristic diagram of the non-recursive filter, and FIG. FIG. 8 is
a block diagram of a conventional example.
DESCRIPTION OF SYMBOLS 1 ... analog input end, 2 ... analog / digital converter, 3 ... digital
source source, 4 ... switch, 5 ... mask clock generator, 6, 7 ... high pass filter , 5a, 7a are recursive
filters, 6b. 7b is a non-recursive filter, 8, 9, ... low pass filter, 10, 11, 12, 13, 24, 25 ... digital /
analog converter, 14-17 ... amplifier ... 18, 19 ...・ Speaker for high-pitched sound, 20.21:
Loudspeaker, 23: Sampling converter, 26 to 29: Delay circuit. → 7 townscaping 4y + Yen ditches
C 乙 (a) (b) Fig. 6 Fig. 7 ■ Round circuit frequency
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