JPS5933927A - Analog/digital converter - Google Patents

Abstract

PURPOSE:To obtain a digital signal with high quality, by operating an A/D converter so as to obtain finally a converted output of the digital signal having a sampling frequency N-times that of the digital signal desired to be obtained. CONSTITUTION:A high frequency component being f1 of the sampling frequency or over is eliminated from the input analog signal at an analog low pass filter 6 and applied to the A/D converter 7 of the next stage. The A/D converter 7 consists of a sample-and-hold circuit section 7a and an A/D converting section 7b, and after the input signal is sampled at the sampling frequency Nf1 at the sample-and-hold circuit 7a, the signal is quantized at the A/D converting section 7b and coded to produce a PCM digital signal, and applied to a digital low pass filter 8, whose frequency at the end of pass band fp' is selected at a frequency <1/2-time the f1 and the block end frequency fs' is selected nearly to 1/2f1. The signal extracted from the digital low pass filter 8 is applied to an thinning out device 9, where the signal is interleaved into the digital signal having the desired sampling frequency f1 and outputted to an output terminal 10.

Description

DETAILED DESCRIPTION OF THE INVENTION Field of the Invention The present invention relates to an A/D conversion device, particularly an A/D converter that converts a temporally continuous analog signal into a digital signal with little quality deterioration and outputs the digital signal. Regarding equipment.

BACKGROUND OF THE INVENTION FIG. 1 shows a block system diagram of an example of a conventional A/D converter. In the figure, an analog signal input to an input terminal 1 is supplied to an analog low-pass filter 2, where the frequency tl number f11 of the digital signal (PCM signal) extracted from an output terminal 4 (to be described later) is - Double laps? 7I2 number -> Frequencies bi equal to or greater than 11 are attenuated. This is to prevent aliasing distortion from being mixed into the output digital signal, and frequencies of 11 or more are attenuated according to the Nyquist sampling constant. The output signal of this analog low-pass filter 2 is
Sampling hold circuit section 3a and A/D conversion section 3b
After being sampled and held in the sampling and holding circuit section 3a, the A/D converter 3 consists of
The signal is quantized and encoded by the D converter 3b, converted into a pulse code modulated (PCM) digital signal, and outputted to the output terminal 4. When the input analog signal is an information signal such as an audio signal or a video signal, the output digital signal is then recorded on a recording medium such as a magnetic tape or a disk.

Here, the analog low-pass filter 2 is configured to have a thorough width-circumference, and in order to make the pass band as wide as possible, IIl passband edge frequency f, is 1■stopband edge circumference S
It is selected very close to the number f5, and the image frequencies f, and f
The frequency band (transition bandwidth) between s and s is narrow.

On the other hand, the attenuation amount by the filter 2 is opposite to the quantization bit number of the A/l) converter 3, and the quantization bit number 1
Since an attenuation of -6 dB is required per bit, for example, if the number of quantization bits is 16 bits, the amount of attenuation is approximately -96 dB.
The attenuation amount of l:l must be 7.

Therefore, for example, when the input analog signal is an audio signal, the passband edge circumferences e and f are 20kHz [
#', since the sampling frequency f1 is about 50 k) Iz, the amount of noise of -96 dB or more is necessarily small in the narrow frequency range of about 5 kHz from frequency f to f, and the analog low-pass filter 2 requires an extremely steep attenuation characteristic. For this reason, conventionally, the analog low-pass filter 2 is often constructed by using a combination of Chebyshev 1.1'' filters.

Problems to be Solved by the Invention Since the analog low-pass filter 2 is required to have a steep damping viscosity with the above-mentioned black sliding door, if you try to meet the attenuation amount +-1, the phase-frequency characteristic will inevitably change. As shown in Fig. 2 (03), the passband edge current wave number f. Within the passband of ±π(
rad), the phase fluctuates greatly. Therefore, the band-limited analog signal supplied to the A/D converter 3 has the disadvantage that it is significantly distorted within the normal band.

In addition, designing this analog low-pass filter is extremely difficult. For example, if you design a filter such as a simultaneous Chebyshev 13th order, there will be variations due to the width of the element, and it is also difficult to adjust the amount of attenuation required. It is difficult to secure
There were many design problems, such as ripples occurring in the passband (deviation from the design value). In addition, in the conventional device, the scale of the analog low-pass filter 2 is large, and the PC
There were drawbacks such as quality deterioration such as the group delay time of the M digital signal being inconsistent.

Therefore, the present invention operates the A/D converter at N times the sampling frequency 11 (N is a natural number of 2 or more), and passes the output digital signal through a digital low-pass filter and a decimator, respectively. It is an object of the present invention to provide an A,/D disguise device that can facilitate the design of an analog low-pass filter and improve the quality of the final output PCM digital signal.

Means for Solving the Problems The present invention is characterized in that the passband edge current wave number is sent to a frequency where the target sampling frequency termination frequency is approximately equal to or less than the sampling frequency t (a natural number greater than or equal to 2). , an analog low-pass filter that attenuates the high frequency components of the analog signal, and the output analog signal of this filter is supplied to the
an A/D converter that performs quantization and encoding at twice the frequency and then obtains a digital signal;
The signal is supplied to the output voltage of the splitting device, and the transient frequency is higher, and the Tado terminal frequency is approximately 1 of the above sampling frequency.
[to the same wave number that is twice or less] The digital low-pass filter that has been fixed and the output digital small signal of the digital low-pass filter are taken out every N times the frequency of the above sampling frequency and sampled at this kind of standard frequency. This device eliminates the drawbacks of the conventional device by developing a decimator that outputs a digital signal.
With Figure and Peng 4 figures; the situation is clear.

Embodiment FIG. 3 shows a block system diagram of an embodiment of the apparatus of the present invention. In the figure, an analog signal that is continuous in time and thread width is supplied to an analog low-pass filter 6 via an input terminal 5.

This analog low-pass filter 6 has the same wave number that is more than double the sampling frequency f1 of the digital signal to be finally obtained in this actual case.For example, N is 2, and the stopband edge yb
The number of pumps f is sampled lI! Assuming that d is equal to the wave number 11, its J-frequency physical properties are shown by the solid line I in Figure 4 (5).
Therefore, the input analog signal is supplied to the A/D converter 7 at the next stage after nine high-frequency components above frequency f1 are removed by the analog low-pass filter 6.

This A/D converter 7 has a sampling hold circuit section 7a.
and A/l) conversion section 7b, and after sampling the input band-limited analog signal at the sampling frequency Nf (2f1 in Japan since N=2) in the sampling hold circuit section 7a, A/D conversion is performed. In the section 7b, for example, the data is quantized by II '3 with the number of embedding bits being 16 bits, and encoded and converted into PCM.
Generate a digital 1za number. This number of rounds fI of this camphor is 2f
The digital signal, which is iE negative in both time and width 1, is supplied to a digital 1 bandpass filter 8.

The digital low-pass filter 8 has a width end frequency fp' and a termination end frequency f5' of an angular wave number below flame ±f1μ.
1 is selected, and here -in;1 is J',
-Tf.

-Then, the thread width - frequency time! '11: is shown in Figure 4, G-A ``g[, HNQ II.

For example, the digital low-pass filter 8 has a angular limited impulse response (F I R) signal expressed by the following difference equation.
It is made up of 4 pieces of r.

1 ( ■ = Σh... In addition, h represents the first filter coefficient.-For example, the passband edge layer wavenumber f,' shown in FIG.
is 20 kHz, and the stopband terminal wave number fs' is 22.05.
kHz, and the digital low-pass filter 8 is capable of obtaining an attenuation of -96 dB between the frequencies f,' and f,'.
It can be configured with an IR digital filter, and in that case, the filter coefficient in the above equation is as shown below. However, H(tl indicates hl, and similarly H(2), H
(3) ,..., H(185) , H(186)
indicates h2゜h31 ''! h11151 h186, and H (nl = H(187-n)) (
n=1.2.3+..., 93).

)1(10)=-0,19287142g-03=[(
(177)H(11)=0.13711511
-03=1((176)H(13)=-0,10766
091E-03=H(174)H(14)=-0,32
001726E-03=H(173)H(15)=
0.51143498E-04=1-1(172)1-
1(16) = 0.39017018E-03
= lI(171)H(17) = 0.298161
16E-04=I-[(170)H(41)=, 0
．． 14927421E-02=)((146)H(42
)=0.1957057.6g-02=H(145
)H(43)=-〇, 12965321Fi-02
=H(144)H(44)=-0,24,498986
E-02=H(143)H(45)=0.9793
5335E-03=H(142)H(46)=0.
29338680E-02=H(1411H(47)=
-o, 531697s 7E-o 3 =x-+(14
0)T-((48)=-0,338313071D-0
2,=I((13,9)[1'(49)==0.53
371290g-04=H(138)H(50)=
0.37669609E-02=H(137)H(5
1)=0.77671176E-03=H(1
36) 1-1(52)=-0,40522388g-0
2=H(135)1-1(53)=-0,]]6330
981E-'02=H134)H(54)=
0.42043891 E-02=H(133)H(
55)=0.26113042E-02=H(l3
2) H(56)=-0,41872268E-02=H
(131)H(57)=-0,36923668B-0
2=H(130)H(58)=0.3965042
5E-02=)i(129)1-1('59)=
0.48498770E-02=H(128)H(6
0)=-0,35037783g-02=)1(127
)H(61)=-0,60507477E-02=)I
(126)H(62)=0.27703853E-
02=)i(125))1(63)=0.7253
2864E-02=H(124)T((64)=-0,
17355734E-02=H(123)H(67)=
0.94625316B-02=1((120)H
(68)=0.13405322B-02=H(1
19) H (69) Knee〇, 10350283E-01=
H(118)H(70) Ni - 0. :U4267
842E-02=H(117)H(71)=0.1
1000999B-01=)-1(116)H(72)
== 0.59065911E-02=f((115
)H(73)=-0,1,1333941E-01=H
, (114)H(74)=-0,880559! dtl
D-02=H(113)H(75, 〇, 1]254
631E-01 =l[112)H(76)=0.
12]61401E-01=HC111)H(77)=
-0,10646204E-01=lI(1,10)H
(78)=-0,]]6035434E-01=1-1
1.09) H(79)=' Q, 93549951
E-02=H(108)1-((80)=0.20
53R495E-01,=1((107)H(81)=
-1'), 71580947E-02=+((106)
T-1, (82) = -Q, 2588], 453E-0
1= )((105)H(83)=0.3691
2166E-02=) [(104.n1-[(84)=
0.32487035B-01=H(103)■(
85)=(117255398E-02=H(10
2) H(86) knee 0.41.278972E-01=
H(101))((87)=-0,10584752B
-01=H(100)H(F18)2 □, 54600
508E-01=1((9,9)H'(893=0
．． 27017994B-01=H(98)1-1(90
)=-0,80275354E-01=H,(97)H
(9])=-tl, 68390612E-01=H(9
'6)H(92)=0.1681403.8E+O
O2l-1(95))1. (93) = 0.4301
6557E+0O=H (94) The band-limited digital signal extracted from the digital low-pass filter 8 is supplied to the decimator 9, where it is decimated into a digital signal with a desired sampling frequency f1, and then output to the output terminal 10. Powered. f, 7, that is, jl from the digital low-pass filter 8
The sampling frequency is the sampling frequency 2f in the A/D converter 7, although the y-output is removed and the band is limited.

are the same, and the target sampling frequency f1 is different. Therefore, the decimator 9 decimates the input band-limited digital signal every other sampling period and extracts only one sample out of two samples, thereby obtaining a digital signal with the target sampling frequency f1. The digital low-pass filter 8 limits the band in order to satisfy the sampling theorem so that aliasing distortion will not be mixed into the output digital signal when the thinning operation is performed using the thinning gain 9.

According to the actual example, the amplitude-frequency customization of the digital low-pass filter 8 becomes as shown by the dotted line (■) in the fourth garden, and the transition bandwidth becomes narrow, but the digital filter can be made with high accuracy. Moreover, since it is digital signal processing, S
There is no deterioration in the N ratio, and the phase-frequency stability is a linear phase as shown by the dashed line (■) in FIG.

Therefore, the A/D converter 7 maintains the sampling frequency f without signal deterioration even when operated at 2f (≠ regularized mantissa).

You can get the digital number of months. On the other hand, since the A/D converter 7 can be operated at a sampling frequency of 2f, the A/D converter 7 is provided at its input stage to eliminate aliasing.
The analog low-pass filter 60) whose amplitude-frequency characteristics are shown in Figure 4 (solid line I at N) has a slope characteristic that is much gentler than that of filter 2. You can obtain the required attenuated pongee. For example, the edge Kata cost f of the Aku no 1 area shown in the 4th park is 30 kl-1z,
q 7;1-11!1” Termination frequency f is 50 kHz
If the frequency range is approximately 20 kHz, which is four times as wide as the conventional frequency range, it will be possible to achieve a frequency range of -96 dB or more.

Therefore, compared to the conventional analog low-pass filter 2, the design of the analog low-pass filter 6 is easier, its scale can be reduced, and deviations from design values due to variations in elements do not occur. Moreover, the phase-frequency characteristic of the analog low-pass filter 6 is approximately a linear phase at least in the frequency band below U, as shown by the solid line ■ in FIG. There is almost no disturbance. Therefore, from the above, compared to the conventional device, the output terminal 10 has
A high-quality digital signal with improved phase-frequency characteristics and reduced aliasing distortion can be extracted.

In the above embodiment, the case where N is 2 has been described, but it goes without saying that it may be a natural number of 3 or more.

Effects As described above, according to the present invention, the A/D converter can be set to N of the sampling frequency f1 of the digital signal to be finally obtained.
Since the operation is performed so as to obtain a converted output of a digital signal with twice the sampling frequency, the slope characteristic of the analog low-pass filter for eliminating aliasing distortion provided on the input side of the A/]) converter is much greater than that of conventional equipment. The design of analog low-pass filters can be made more gradual, therefore compared to traditional analog low-pass filter designs,
It is much easier to manufacture, the scale of the filter can be made smaller, and the occurrence of r deviation from the design value due to element variations can be almost eliminated. Since the frequency characteristics are approximated by a linear phase, phase distortion can be removed, and as a result, it has the advantage of being able to output a higher quality product signal than the conventional method.

[Brief explanation of the drawing]

Fig. 1 is a diagram showing the block system characteristics of an example of a conventional device, and Fig. 2 is a diagram showing the width-frequency-temporal-1-phase-circumferential 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, and FIG.
FIG. 2 is a diagram showing frequency characteristics 1 phase-frequency discrimination. 1.5...Analog signal, input terminal, 2,6...Fist analog low-pass filter, 3,7-..., A/D conversion 4 spindles. 4.10...φ tactile signal output leprosy, 8.Fist digital low-pass filter, 9...decimator. Figure 1 Figure 2 Figure 3 'h 7b Figure 4 fp' ty2fI voice J4

Claims (1)

[Claims] In an A/D conversion device that samples an analog signal at a target sampling frequency and then converts it into a digital signal that is discrete in both time and amplitude, the passband terminal wave number is approximately one or more times the sampling frequency or more. , and the stopband end-office wave number is selected to be a frequency approximately equal to or less than the sampling frequency (N is a natural number of 2 or more) to attenuate the high frequency components of the analog signal. An analog low-pass filter and an A/D conversion where the output analog signal of the analog low-pass filter is supplied, samples it at a frequency N times the sampling frequency, and then performs quantization and encoding to obtain a digital signal. and the output digital signal of the A/D converter is supplied, the passband end layer wave number is selected to be less than approximately 1 times the sampling frequency, and (1) the toe end frequency is approximately equal to the sampling frequency. A digital low-pass filter selected to have a frequency equal to or lower than the above-mentioned sampling frequency, and a digital signal output from the digital low-pass filter at every frequency N times the sampling frequency, and a digital signal that is not sampled at the sampling frequency. An A/D variable device characterized by comprising a thinning chair for outputting data.