JPS5933928A - Digital/analog converter - Google Patents

Abstract

PURPOSE:To improve the quality of an output analog signal, by converting the sampling frequency of a digital signal to high, and passing it through a D/A converter. CONSTITUTION:A PCM digital signal sampled at the sampling frequency f1 and quantized and coded is inputted to an interpolating device 6, and after the signal is converted into a digital signal having the sampling frequency converted substantially N-times, and applied to a digital low pass filter 7 having the pass band extremity frequency fp' being <=1/2-time the f1 and the block end frequency fs' being f1/2. A band limit digital signal extracted from the digital low pass filter 7 is applied to a D/A converter 9. The D/A converter 9 consists of a D/A converting section 9a and a sample-and-hold circuit section 9b and is constituted so as to convert the digital signal having Nf1 of the sampling frequency into an analog signal. An analog low pass filter 10 eliminates the frequency component of >=3f1/2 included in the analog signal extracted from the D/A converter 9.

Description

[Detailed Description of the Invention] The present invention relates to a D/A converter, and particularly to converting a digital signal that is discrete in both time and width into an analog signal with less deterioration in quality. This invention relates to an output l)/A conversion device.

BACKGROUND OF THE INVENTION FIG. 1 shows a block system diagram of seventeen examples of a conventional D/A converter. In the figure, a digital signal input to the input terminal 1 whose time and amplitude are both discrete, such as a pulse code modulation signal (PCM signal), is composed of an I)/A conversion section 2a and a sampling hold circuit section 2b. /A converter 2. The D/A converter 2 operates at the sampling frequency f1 of the input digital signal, converts it into an analog signal that is continuous in both time and amplitude, and outputs the analog signal. The output analog signal of this 1]/A transformer 2 contains a frequency component that is more than one time the original frequency f1, so according to the sampling theorem, the average frequency f s O) is less than twice the original frequency f1. The signal is outputted to an output terminal 4 through an analog low-pass filter 3 for extracting only the frequency components.

Here, in order to make the pass band as wide as possible according to the sampling theorem, the analog low-pass filter 3 is
The fl passband edge frequency f, is selected to be close to the blocking edge frequency f3, and its amplitude-period δν frequency characteristic satisfies the condition φ, as shown in the second garden, 1 f, < f8≦■ fn& has been made to do so. Therefore, the frequency band between the passing edge frequency f and the 15h stop edge frequency f, that is, the transition band width becomes narrow. For example, in the case of audio signal transmission, the above-mentioned 101 Shojo edge frequency fp is about 20 kl-1z, and the sampling frequency f.

is about 5 Q kHz, so the transition bandwidth is about 5 Q kHz.
It is only about kHz. On the other hand, the attenuation amount of the filter 3 is -96 dB compared to the number of quantization bits of the digital signal when the number of simplification bits is 16 bits. Therefore, the analog low-pass filter 3 is required to have an extremely steep attenuation characteristic, so conventionally a simultaneous Chebyshev 11th to 13th order
A filter of the order of 5- is often designed and applied.

Problems to be Solved by the Invention The analog low-pass filter 3 is required to have a simple attenuation characteristic based on the above-mentioned sigma. It is difficult to ensure the integrity of the filter, and it is necessary to allow the occurrence of ripples within the passband in order to ensure the amount of attenuation that exceeds the design value. Due to its large size, there were many design and manufacturing problems. Furthermore, in order to obtain the above-mentioned steep attenuation characteristic, the phase-frequency characteristic of the analog low-pass filter 3 will inevitably have a passband end wave number f, as shown in FIG. ±π within the band (
rad), and this could not be suppressed. For this reason, the analog 4 having a variable 1ψ has a disadvantage in that the quality of the signal, especially regarding the position λ(), deteriorates.

Therefore, the present invention (the sampling frequency f1 of the digital signal
E, ヂ6〈By converting and filling in D/A conversion 4,
My father decided not to use analog low-pass filters, but the design of the JJ/AX replacement neck (4, 1st order 1
The present invention (each sample value of 4-input digital 4-member consideration is N-1 (where N is a natural number of 2 or more) ) by inserting a zero point, the sampling frequency f is substantially converted to N times and a digital signal is output.The output digital signal of this interpolator is supplied to 6 interpolators, and the edge circumference of 1 passband is The frequency of the sampling frequency f is selected to be less than 'fVz in terms of the sampling frequency f, and 1) 14+f-jjrjE frequency is set to be 7 times or less of the sampling frequency 11 in actual terms. The digital 1-pass filter and the output of this digital low-pass filter are set to the sampling frequency a, 7
(2) The shortcomings of the conventional device are eliminated by using a 1/A converter that operates at N times the frequency and converts it into an analog signal and outputs it. An example will be explained with reference to FIGS. 3 to 5.

Embodiment FIG. 3 shows a block system diagram of an embodiment of the apparatus of the present invention. In the figure, a PCM digital signal that has been sampled at a sampling frequency of 11, then subjected to amplification and 90 coding is supplied to an interpolator 6 via an input terminal 5. This interpolator 6 performs N-1 (N-1 interpolators, however, N
is a natural number greater than or equal to 2), and thereby outputs a digital signal with a one-vote frequency Nf1 in which the sampling frequency is actually changed by a factor of N. Therefore, if the input digital signal having the frequency spectrum shown in pattern 4 is subtracted by 2 from the above N, then the interpolator 6
As shown in FIG. 4 (B1), the signal is converted into a digital signal with a sampling frequency of 2f1 having a frequency fJ9'@spectrum of 4J'', and is outputted by the digital filter 7.
supplied to

The digital signal with sampling frequency 2f1 is expressed as θ'8 in Fig. 4.
) shows a frequency spectrum, and since there exists a signal/component below the frequency ζ, there is no problem in removing the larger frequency component.

Therefore, the digital low-pass filter 7 is +k l:
The V=j-frequency special characteristic is shown in Figure 5 (indicated by a chain I), and the number f' at the edge of the Go-1 area is ↓ times the sampling frequency f1 2 4- The sampling frequency of 211 is around the tacital scar code?
F! 'a Ieu< output f le. f, f El,
Figure 454 (81, (C''l) (1) The diagonal line in the frequency spectrum indicates that the yeast ratio frequency is 2f, which is a digital signal.

This digital low-pass filter 7 (well, B. Rie is the first impulse response (J
i' I R) It is configured as a digital filter.

■ -Σh・・X・nl− However, in the above formula, x and y are respectively at time nT
denotes the value of the input digital signal and the value of the output digital signal at n n sampling periods), and denotes the first filter coefficient. - For example, if the passband end station wave number f' shown in Fig. 5 (8) is 20 kHz, and the stop band end office wave number fS is 22 kHz.
．． 05 kHz, and their frequencies a f,' and fS
The digital (legal pass filter 7) that can obtain an attenuation of -96 dB in the transition bandwidth between The coefficient , becomes the following formula. However, l-1 (11 indicates hl, and similarly H(21, H(31, . , 'H(185)
.

H(186) is h2. h3.゛ νh185・h18
6 and 1 and H 6″n) = 14 (187-m
) becomes (rn==1, 2, 3 r-rI-1(
z) = 0.78957020E-04=H(1
85) H(3)=-0,46632811E-04=H
(11'+4)H(4)=-0,30974154Fi
-03=H(183)H(5)=-0,4242828
4E-03-2)((182)H(6)=-0,18
672901-03=H(181)H(7)=
0.13627916E-03=H(180)H(8
) = o, 1120842fJ-03=
H(179)H('9)=-0,17278915
B-03= )((178) t1(10)=-0,19
287142E-03=H(177)H(14)=-0
, 32001726-103=)((173)H(3
3)=0.13764823E-02=H(154
)H(34)=0.29210552B-03=
H(153)1-I(35)=-0,15065504
E-02=H(152)H(36)=-0634615
83E-03=1-1(151)H(37)=0.
'15812291B-0'2=r-1(150)H(
38) = 0.10334401-02 =H(149
)H(39)=-0,15823867E~02=H(
14B) H(40)=-0,1,4789228g-0
2=H(147)1-((41)=0.1.492
7421E-02=H(146)H(42)=0.
19570576E-02=H(145)to1(4phantom
=-0,12965321E-,02=H(144)H
(44)=-0,2'4498986E,02=H(,
143) T-I(45) = 0.979353
35B-0'3=l((142)to1(46)=
0.29338680B-02=l((14
1,) t1(47)=-0,53169787E-03
=H(140)1-1(48)=-0,3383130
7g-02=H(139)H(51)=0.776
71176E-0371-((136)1((52)=
-0,40522388E-02=1((135)H(
53),,=-0,16330981E-02=H(1
34) H(!54) = 0.42043891E-,
02=H(133)1-1(55)=0.2611
3042hi-02-1 ((132)H(56)=-0
,4187226RE-02=H(131)H(57)
=-0,36923668g-02=H(1,30)to1(58)=0.39650425E-02
=H(129)H(59)=0.48498770
E-02=)I(128)H(60)=-0,3503
7783E-02=H(127)1-1(61) Knee〇, 60507477E-02=H(126)H(62)
= 0.27703853E-02=H(125)[
((63) two 0.72532864F3-02=i-
1(124)H(64)=-0,17355734E-
02=H(123)H(65)=-0,8408951
,9T 102=l-1(1,22)H(66)=
11.37327550E-03=H(121)to1(
67)=0.94625316g-02=1
-1(120)+((68)=0.1340532
2E-02=lI(119)1-T(69)=-0,1
0350283g-01=H(118)II(70)=
-Ll, 342678421 C02=l((1,17
)Ll(71)=0.11000999E-01=
I((116)H(72)=0.590659
11E-02=l((11,5)H(73)=-0,1
1333941E-01=H(114)H(74)=-
0,880559580-02=t4(113)H(7
5) = 0. ],'12546311see01=l(
(112)1-+(76) = 0.1216140
1B-0 to H(1,11,)1-1(77)=-0,1
0646204E-01=H(110)1-1(78)
=-1) 16035434g-01=1-1(109)
H(79) = 0.93549951. B-02
Two 1 (108)) - 1 (80) = 0.205
384951D-01=)I(107)jl(81)nee071580947E-02=l((106))-1
(82)=-(125881453E-01=H(10
5) To1(83) = 0.3691216
6g-02=H(104)II(84)=0.
32487035J"3-01=H(103)]-+1
85)=0.172553981B-02=I(
(102)1i(86) = -0,412789
721 code 01=H(101)1-1(87)=
-0,105847521D-01-t(('100
)1((88)=0.54600508E-01=
+1(99)+1(89)=0.2'701799
．． 4B-01=H(9B)H(90)=-0,8027
5354E-01=1-1(97)H(91)=-0
, 68390612g-01=1-1. (96)H(9
2)=0.16814038E+OO=H(95
)H(93)=+143016557g+0O=H(
94) The band-limited digital signal extracted from the digital low-pass filter 7 is supplied to the D/A converter 9 via the terminal 8. The D/A converter 9 is r)/A modification section 9a.
and a sampling hold circuit section 9b, and is configured to convert a digital signal of sampling frequency Nf, (2f1 in this case) into an analog signal, as shown in FIG.
A digital signal with a sampling frequency of 2f and a spectrum shown in q is converted to analog (D
It outputs an analog signal that covers the frequency spectrum as shown in +.

The analog low-pass filter 0 is provided to remove frequency components of 711 and higher shown in FIG. 4(D) contained in the analog signal taken out from the D/A converter 9, and is As shown by the solid line ■ in 5), the type width-frequency characteristic is such that the passband end-office wavenumber f is above and the stopband end-office wavenumber f is substantially (N+1)f or less S
2 1 frequency (here, N=2, so f, = f1) is selected as 1st generation. This results in
An analog signal having a frequency spectrum shown in FIG. 4(B) is extracted from the analog low-pass filter 0 and output to the output terminal 11.

According to this embodiment, the width of the digital low-pass filter 7 is -
The frequency characteristic (Figure 5) is shown by a dashed line I, and although its transition bandwidth is narrow, the ticle filter can be constructed with high precision, and since it is digital signal processing, there is no deterioration in the S/N ratio, and the transition bandwidth is narrow. Its phase-frequency characteristics are shown in Figure 5 (G).
) is a perpendicular phase indicated by a dashed line ■, and there is no deterioration of Hakugo product trade due to phase disturbance. Therefore, the D/A converter 9 is supplied with a digital signal (g) from the digital low-pass filter 7 without any deterioration.

Other power, 1. )/A converter 9 has a sampling frequency of /? Since the digital signal of 12f1 is converted into an analog signal, the amplitude-frequency characteristic of the analog low-pass filter 10 provided at the output stage to remove the young frequency component is shown in FIG. 5 (N As shown by the solid line ■, since the transition bandwidth is extremely wide compared to that of the conventional analog low-pass filter 3, it is possible to obtain the required attenuation with an extremely gentle slope characteristic. ), the passband edge frequency f, shown in
When the stopping frequency f5 is 50, T(z is about 7, about 2
It is sufficient to obtain an attenuation amount of -96 dB or more with a transition bandwidth about 4 times as wide as that of the conventional method.

Therefore, compared to the conventional analog low-pass filter 3, the design of the analog low-pass filter 10 is easier, its scale can be reduced, and deviations from design values due to variations in elements hardly occur. Moreover, in the lower frequency band, the phase is approximately linear, and there is almost no phase distortion within the ultimately necessary band.

Therefore, from the above, the phase-frequency characteristics of the output terminal 11 are improved compared to the conventional device, and as a result, the quality of the analog signal is greatly improved.

Note that although the analog low-pass filter 10 is provided in the above embodiment, when the analog signal is an audio signal, the sampling frequency 11 is about 45 kl-1z,
The analog low-pass filter 10 removes non-defective frequency components having frequencies of 11 or higher, which are outside the audible frequency band. Therefore, if the device is to be f+M-friendly, it is not necessary to use the analog low-pass filter 1o.

Application example When the input digital signal and output analog signal are primary signals such as video signals and audio signals, the output digital signal of the digital low frequency filter 7 can be recorded on a recording medium such as a magnetic tape or a disk. . In this case (well, the digital signal obtained by reproducing this recording medium 2 is connected to the terminal 8 provided in the reproducing device)/A converter 9.
By supplying the signal through the output terminal 11, a reproduced analog signal of the pot price can be taken out from the output terminal 11 compared to the conventional case.

In addition, although the case where N is 2 was explained in the above-mentioned example, it goes without saying that it may be a natural number of 3 or more.

Effects As described above, according to the present invention, since digital-to-analog conversion is performed by multiplying the sampling frequency f1 of the input digital signal by N times, a filter is provided on the output side of the L/A converter to remove unnecessary frequency components. When an analog low-pass filter is provided, its transition bandwidth can be made much wider than before.
Therefore, compared to conventional analog low-pass filters, it is much easier to design and manufacture analog low-pass filters, the scale of the filter can be made smaller, and deviations from design values due to element variations can be almost eliminated. The circuit can be simplified, and the phase-frequency characteristics of the analog low-pass filter in the required band of the output analog signal can be approximately made into a linear phase, so phase distortion can be removed. It has many features such as being able to output analog signals.

[Brief explanation of the drawing]

1 is a block diagram showing an example of a conventional device; FIG. 2 (At) and (B) are diagrams showing the finger width-frequency characteristic 2 phase-frequency characteristic of the analog low-pass filter in the device shown in FIG. 1, respectively; Fig. 3 is a block system diagram showing one embodiment of the device of the present invention, Figs. Figures (3) and (B) are diagrams showing the wave width-frequency characteristics 9 phase-frequency characteristics of the analog low-pass filter and the digital low-pass filter in the apparatus shown in Figure 3, respectively. ], 5... Digital signal Iruka Mizuko, 2,9...D
, /A conversion g=13. toee0 analog f-range filter, 4.11・fist・analog signal output terminal, 6・interpolator, 7・digital low-pass filter. 5th cause Figure 2 Figure 3 Figure 5 Procedural amendment September 2, 1982 1, Indication of case 1982 Patent application No. 143627 2, Title of invention D/A converter 3, Make amendment Patent Applicant Address Kumo 221 3-12 Moriya-cho, Kanayō-ku, Yokohama-shi, Kanagawa Prefecture Name (432) Victor Japan Co., Ltd. Representative Director and President Shishi 3 - Part 4, Agent 5, LI attached to amendment order Voluntary amendment 6, Claims column for the specification subject to amendment. 7. Amend the claims section for the description of the amendment as shown in the attached sheet. 2. Claims [(1) 11) Both ripples and amplitudes are discrete/, and each ripple of the input digital No. I'm-1 pieces between servings (N is a natural number greater than or equal to 2)
An interpolator that outputs a digital signal obtained by substantially converting the sampling frequency f1 by N times by inserting the zero point of
The output digital signal of the interpolator is supplied so that the passing weave frequency is selected to be substantially less than 1/2 of the sampling frequency 11, and the blocking weave frequency is selected to be substantially less than the sampling frequency [1]. a digital low-pass filter selected to be 1/2 or less, and the output digital signal of the digital low-pass filter is operated at a frequency N times the sampling frequency [I] to convert it into a 7 analog signal and output it. A D/△ conversion device that takes a 12-inch configuration from a 7'A transformer. (2) The D/Δ conversion device according to claim 1, wherein the input digital signal is a signal obtained by subjecting an A-dio signal to a log-digital conversion. (3) In a device that converts a digital signal with a sampling frequency f1 whose time and amplitude are l111 to an ablog signal, there is N between each ripple value of the input digital signal.
- an interpolator that outputs a digital signal obtained by substantially converting the sampling frequency 11 by N times by inserting one (N is a natural number of 2 or more) zero points; The supplied passing weave frequency is selected to be substantially less than 1/2 of the sampling frequency [1, and the blocking weave frequency is selected to be substantially less than 1/2 of the marking frequency [1]. 2 A digital low-pass filter selected to be f8 or less and the output digital signal of the digital low-pass filter are operated at the frequency of the sampling frequency 11 (N) to convert it into an analog signal and output it. /A converter and the output analog signal of the D/A converter are supplied so that the passing weave frequency is substantially 1/1 of the sampling frequency [1].
The frequency is selected to be twice or more, and the blocking edge frequency is actually selected to be less than (N+1>/2 times the sampling frequency f1), and the frequency of the target analog signal is selected and the -C output is overridden. A D/A converter featuring an analog low-pass filter configuration.

Claims (1)

[Claims] (11) In an apparatus for converting a digital signal into an analog signal with a sampling frequency f, which is both discrete in time and type width, N-1 (however, an interpolator that outputs a digital signal obtained by substantially converting the sampling frequency f1 by N times by inserting zero points (N is a natural number of 2 or more); and a passband terminal station to which the output digital signal of the interpolator is supplied. The wave number is the above sampling frequency f1
a digital low-pass filter whose frequency is selected to be substantially less than 1 times the sampling frequency f1 and whose rejection cutoff frequency is selected to be substantially 7 times or less than the sampling frequency f1; and an output digital signal of the digital low-pass filter. The above sampling frequency 1
D characterized in that it is comprised of a D/A converter that operates at a frequency of N times 1 and converts it into an analog signal and outputs it.
/A conversion device. (2) The D/A converter according to claim 1, wherein the cough input daisy signal is a signal obtained by analog-to-digital conversion of an audio signal. (31 In a device that converts a digital signal with a sampling frequency f1 that is discrete in time and type width into an analog signal, there are N-1 between each sample value of the input digital signal (N-1 (N is a natural number of 2 or more)) An interpolator outputs a digital signal obtained by substantially converting the sampling frequency f by N times by inserting a zero point, and the output digital signal of the interpolator is supplied so that the passband terminal wave number is unified as described above. frequency f,
a digital low-steep filter whose frequency is selected to be substantially less than 7 times the sampling frequency f1, and whose blocking one end station wave number is selected to be substantially 7 times or less the sampling frequency f1; An I)/A converter operates at a frequency N times the sampling frequency f1 to convert the output digital signal into an analog signal and outputs the analog signal, and the output digital signal of the I)/A converter is supplied. The passband end station wave number is the above-mentioned Kumoto conversion frequency vi, f. 2. The stopband edge frequency is substantially N+1 of the sampling frequency 11.
[)/A f press device characterized by comprising an analog low-pass filter that selects a frequency and outputs a target analog signal which is selected below the frequency range.