JPS5820491B2 - Double-sided frequency division multiplexing signal transmitter/receiver for dial signals - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION The present invention provides a method for frequency-division multiplexing a dial signal with a call signal by amplitude modulation in a carrier telephone terminal equipment, or for extracting and restoring the dial signal from multiplexed communication information. A double-sided frequency division multiplex signal for dial signals (hereinafter referred to as DSB- for dial signals) using the out-of-band transmission method of
This relates to the miniaturization of transmitting/receiving devices (called FDM signals).

Conventional transmitters of this type are turned on and off by dial pulses when the number of multiplexed channels is N, and the oscillation frequency is (fc+4Xn) kHz (where fc is, for example, 3.8
5kHz or 3.825kHz frequency, n
is N consecutive integers) are provided for N channels, that is, N circuits, and each output is band-limited by N band-pass filters with a center frequency of (fc+4Xn) kHz to prevent harm to a telephone signal. After limiting the bandwidth to a value that does not give a signal, the sum of N channels is taken and frequency division multiplexed to obtain a DSB-FDM signal for dial signals.

In addition, in the receiving device, a double-side band signal for the Daikiru signal (hereinafter referred to as D for dial signal) is frequency-division multiplexed with the call signal.
In order to extract the SB signal), the center frequency is set to (f
c+4Xn ) kHz bandpass filters for N channels, that is, N circuits, were provided, and the output of each DSB signal for the dial signal was detected by an envelope detector and restored to a dial pulse.

Therefore, the number of channels that a bandpass filter can multiplex! Since the same number as N is required, there is a drawback that the device becomes large when the number of multiplexing is large.

The present invention is directed to band limiting of DSB signals for dialing signals, or DSB signals for dialing signals that are frequency division multiplexed with speech signals.
The device is characterized by using a comb filter as a film for extracting the SB-FT)M signal, and its purpose is to downsize the device, which will be explained in detail below.

FIG. 1 is a block diagram showing one embodiment of the present invention, which performs dial pulse transmission processing when the number of channels is N-12.

In the figure, 1.1 to 1.n-1.12 (the entire group is represented by 1.n) are input terminals that receive dial pulses of the n-th channel, respectively.

Here, in the following explanation, consecutive N pieces (12 pieces)
The integer n of S is treated as an integer from 1 to 12.

2.1 to 2.n-1. connected to this input terminal 1.n.
12 (represented by 2・n when indicating the entire group) is an oscillator that is turned on and off by dial pulses, and its oscillation frequency is 4X mkHz↓ (m is 0) for each channel.
~N-1 natural numbers, where m=n 1 corresponds to each channel) are assigned.

Further, 3 is an adder for calculating the sum of the outputs of each oscillator 2.n, 4 is a comb filter, and 5 is a D/A converter.

Furthermore, 6 is a frequency converter, 7 is a local oscillator, and both constitute an analog mixer with a conversion frequency of 104.15kHz, and 8 is a dial signal DS.
This is an output terminal that sends out a B-FDM signal.

Here, the aforementioned oscillators 2.n, adder 3, and comb filter 4 are circuits that process signals in a digitized manner, and their sampling frequency is 96 (=8X12=8X
N) kHz is used.

Here, when the dial pulse is sent at 10 PPS, that is, 10 pulses/second, the pulse width T of the dial pulse at the input terminals 1 and n is -millisecond.

The output spectrum of oscillator 2·n when 30 TP=33 milliseconds is shown in FIG.

In the figure, a indicates the input pulse of input terminals 1 and n,
b shows the output spectrum of oscillator 2·n.

As shown in the figure, the output of oscillator 2.n is 4×
Since it is an on/off signal of mkHz, its spectrum is attenuated by about 52 dB at a frequency 40 kHz away from the center frequency of 4 x mkHz, that is, a frequency where a phosphorescent channel exists.

Therefore, when the output signals from oscillators 2.1 to 2.12 are added, the influence of the contact channel on the desired channel is sufficiently attenuated with respect to the desired channel.

However, (4Xm+0.3) ~ (4Xm+3.4
) kHz band is for later carrying frequency-divided speech signals, and dial signal components within this band must be attenuated by a predetermined amount.

This filtering is performed by a comb filter 4.

FIG. 3 shows how the dial signal is band-limited by the comb filter 4.

In the same figure, a shows the output spectrum of each oscillator 2 and n, and a 4XmkHz intermittent signal is converted to 96kHz.
Since sampling is performed at z, the spectrum of the dial signal also appears at the target position centered on 48 kHz.

Also, in the same figure, the signal is frequency-division multiplexed by adding these signals in the adder 3, and the figure C is the frequency characteristic of the comb filter 4, and d is the frequency division signal whose band is limited by the comb filter 4. Figure 2 is a multiplexed dial signal spectrum.

The comb filter 4 used here has a 96kHz power as described above.
operates at a sampling frequency of z, and the center frequency is 4, respectively.
×1KH2 (l is O~2×N, that is, a natural number from 0 to 24)
It is a dicicle-type filter with a passband of 1, and its detailed frequency characteristics are shown in FIG.

As clearly shown in the figure, this comb filter has 4
In addition to having a passband of 2 × Δf (however, Δf in the + direction when l = 0 and Δf in one direction when 1 = 24) Hz with the center frequency at XlKHz, it also has a passband of (4i + 0.3).
It has an attenuation range from (4XA+3.4)kHz.

Further, the amount of attenuation in each of these passbands is 3 dB or less, and an amount of attenuation of 40 dB is given in each attenuation band.

Here, since the DSB signal for the dial signal is frequency division multiplexed at a level approximately 30 dB lower than the call signal, the influence on the call signal, including the 40 dB attenuation in the attenuation range of the comb filter 4, is less than -70 dB.

Moreover, the bandwidth 2×Δf of the passband of the comb filter 4 is a parameter that determines the distortion rate when a dial signal is received, and in order to suppress the distortion rate to ±3% or less, 200H7 is selected.

As mentioned above, oscillators 2.n, adders 3. The comb filter 4 operates digitally, and is composed of a recursive digital filter with a delay unit of □×24-96×103 0.25 milliseconds1, and its output is It is converted into an analog quantity by the D/A converter 5.

If the center frequency of the dial signal DSB signal is, for example, the first channel, the output of the D/A converter 5 is OHz, so it is converted to 104.15 kHz by the frequency converter 6.

The output from the frequency converter 6 is a diamond signal DS.
It becomes a B-FDM signal and is sent to the output terminal 8, and its frequency band falls within the basic group region.

This dial signal DSB-FDM signal is summed with the frequency division multiplexed speech signal in the area of the basic group.

゛The state of this frequency conversion is shown in Fig. 5.

In the figure, a indicates the output spectrum of the D/A converter 5, and the figure shows the output spectrum of the frequency converter 6.
D for dial signal frequency converted at 4.15 kHz
The spectrum of the SB-FDM signal is also shown in Figure C.
d shows the spectrum of a speech signal multiplexed with 12 channels by frequency division multiplexing, and d in the same figure shows the spectrum of the multiplexed speech signal indicated by C and the dial signal DSB-FDM signal indicated by b which are further frequency-divided. Multiplexed communication information.

This shows the spectrum.

As explained above, in the prior art, as many band-limiting filters as the number of multiplexed filters are required, but in this embodiment, band-limiting is possible with only one comb-shaped filter using a recursive digital filter.

In addition, the comb filter based on the cyclic digital filter can be easily realized in a small size by increasing the delay amount of the delay line that is a component of the comb filter, so it has the advantage that the dial signal DSB-FDM signal transmitter can be made smaller. be.

Above, an embodiment in which the present invention is applied to a transmitting device has been described in detail. Next, a case in which the present invention is applied to a receiving device will be described.

FIG. 6 is a block diagram showing one embodiment thereof.

In the figure, reference numeral 9 denotes an input terminal, from which communication information in which a multiplexed call signal and a dial signal DSB-FDM signal are added arrives.

As mentioned above, the center frequency of each dial signal DSB signal is (108,15-4X n )kHz.

Further, in the figure, 10 is a frequency converter, 11 is a local oscillator, and 12 is an A/D converter.

The communication information is frequency-converted in the frequency converter 10 according to the conversion frequency of 104.15 kHz generated by the local oscillator 11. Therefore, the center frequency of each dial signal DSB signal is 4 x m kHz.

The communication information frequency-converted in this way is sent to the A/D converter 12.
The signal is sampled at a sampling frequency of 96 kHz and converted into a digital signal.

Next, in FIG. 2, 13 is a digital comb filter, and 14 is a 24 (-2XN) point discrete Fourier transform circuit (hereinafter referred to as a DFT circuit).

15.1 to 15.12 and 16.1 to 16.12 (hereinafter represented by 15.n and 16.n) are DFT circuit 1.
This is an output terminal that outputs the real part or the imaginary part of the 4 kHz signal component of the 4 kHz output signal.

As shown in FIG. 7, the comb filter 13 has almost the same characteristics as the comb filter 4 described above.

That is, (4Xl+0.3) ~ (4xdx3.4)kHz
(7) Attenuation range and (4×e-Δf) to (4×l+Δf)
They are the same in that the attenuation in each passband is 3 (IB), and the attenuation in the attenuation range is 7.
The only difference is that it is 0 dB.

, the communication information converted into a '-magittal signal by the A/D converter 12 is given a large amount of attenuation (70 dB) to the speech signal part in the comb filter 13. The only signal that is sent is the DSB signal section for dial signals.

In this way, the frequency component of the signal given to the DFT circuit 14 has a center frequency of (n-1) x 4 kHz, a bandwidth of only ±Δf, and a time width of the unit of Fourier transformation of (1/: 96) x 24 milliseconds, so
This is an integral multiple of the period of the dial signal DSB signal.

Therefore, the output of the DFT circuit 14 represents the spectrum of the output signal of the comb filter 13 with sufficient accuracy.

Furthermore, in the same figure, 17.1 to 17.12 (hereinafter 17.
・Represented by n) is an arithmetic circuit that calculates the square root of the sum of squares, that is, the absolute value, and is 18.1 to 18.12 (hereinafter referred to as 18
・Represented by n) is its output terminal.

From the output terminals 15.n and 16.n of the DFT circuit 14, the real part or imaginary part of the 4 kHz signal component is sent to the arithmetic circuit 17.n, where its absolute value is calculated.

The calculation result is output from the output terminal 18.n and becomes the dial pulse of the nth channel.

As described above, according to the present invention, it is only necessary to prepare a single comb filter in the receiving device as a filter for separating the communication signal from the speech signal, and the effect on the miniaturization of the device is extremely significant. .

Further, it is more effective to use a fast Fourier transform circuit as the DFT circuit.

[Brief explanation of the drawing]

Fig. 1 is a block diagram showing an embodiment of the present invention in a transmitter, Fig. 2 is a diagram for explaining the spectrum of a dial pulse, and Fig. 3 is a diagram for explaining the state of band limitation by a comb filter. , FIG. 4 is a diagram showing the characteristics of the comb filter, FIG. 5 is a diagram for explaining the frequency conversion, and FIG. 6 is a diagram showing an embodiment of the present invention in a receiving device. The Tanck diagram in FIG. 7 is a diagram showing the characteristics of the comb filter. 1.n + 9...input terminal, 2.n...oscillator,
3... Adder, 4, 13... Comb filter, 5.
...D/A converter, 6,10...Frequency converter, 7,
11... Local oscillator, 8, 18.n... Output terminal,
12...A, /D converter, 14/DFT circuit, 15
・n, 16・n...DFT circuit output terminal, 17・n
...Arithmetic circuit.

Claims (1)

[Claims] 1. The number of multiplexed channels is N1. The carrier frequency is (f
c + 4 An oscillator is provided for each channel, and after obtaining a total signal for N channels, a comb filter is used to limit the band, and this output is frequency-converted by an analog mixer to dial the carrier frequency (fc+4Xn) KHz. The receiving device is configured to generate a double-sided frequency division multiplexed signal for signals, and after obtaining communication information in which the multiplexed call signal and the double-sided frequency division multiplexed signal for dialing signals are added, A comb filter is used to remove the call signal, and this output is further converted by a discrete Fourier transform circuit, and the output corresponding to each channel is restored to a dial signal by N arithmetic circuits that calculate the absolute value. A double-sided frequency division multiplex signal transmitting/receiving device for dial signals, characterized in that: 2. The double-sided frequency division multiplexing signal transmitting/receiving device for dial signals according to claim 1, characterized in that a cyclic digit filter with a selected delay amount of a delay line is used as the comb filter. 3. The comb filter according to claim 1, characterized in that the center frequency of its passband is 4 x 1 KH2 (l is a natural number from O to 2 x N) and the sampling frequency is 8 x N kHz. Double-sided frequency division multiplexing signal transmitter/receiver for dial signals.