RU111369U1 - DIGITAL COMMUNICATION SYSTEM - Google Patents

Abstract

 A digital communication system consisting of a source of an analog speech signal (PC), a pulse generator, a time delay unit and series-connected signal sampler, a half-wave signal rectifier, an AIM-1 to AIM-2 modulation converter, an encoder, a parallel code converter in serial, signal amplifier, and the first input of the code converter is connected to the output of the sampler, and the pulse generator is connected to the second input of the sampler directly and to the second input to the AIM-1 to AIM-2 converter through a time delay unit, and on the receiving side, from a series-connected amplifier, pulse regenerator, serial code to digital signal converter (DS) to parallel, a decoder, a signal multiplier, a low-pass filter, and from the recipient of information, the second input of the signal multiplier connected to the first output of the code converter, the clock input of which is connected to the second output of the pulse regenerator, characterized in that it is additionally introduced into On the receiving side, a 2-fold divider of the PC frequency band, through which the PC source is connected to the low-frequency input of the sampler, and on the receiving side, a 2-fold frequency band doubler, through which the receiver of information is connected to the low-pass filter, and the bit width of the encoder and decoder increases from 8 to 13 , and code converters from 8 to 14.

Description

The utility model (PM) refers to digital systems with a pulse-code modulation (PCM) used to transmit analog speech signals (PC).

Known digital communication systems described in the sources:

1. Patent of the Russian Federation on PM No. 92586. Multichannel analog-to-digital signal converter / A.A. Volkov, G.V.Dolganova, O.E. Zhuravlev. Priority November 13, 2009

2. RF patent for PM No. 102443. Multichannel digital-to-analog signal converter / A.A. Volkov, G.V.Dolganova, O.E. Zhuravlev. Priority 10/20/2010

3. Neyman V.I. Systems and networks for data transmission in railway transport. - M .: Route, 2005. - S.124-126.

In terms of technical nature, the PCM system is closest to PM, the ADC of which is described in the first source, and the DAC in the second, which for this reason are taken as its prototype. The third source describes an analogue of the system.

Prototype - the system consists of a source of an analog speech signal and an analog-to-digital converter (ADC) on the transmitting side, and a digital-to-analog converter (DAC) on the receiving side. The ADC includes time-dependent speech signal sampler, a compressor that compresses sample levels, a sample expander, an encoder that converts sample levels into a binary digital signal. The main blocks of the DAC are a decoder, an expander at its output, which expands the levels of the restored samples to the reciprocal of their compression in the compressor. A compressor and an expander, called a compander, are non-linear devices.

A digital telephone channel with such a non-linear (logarithmic and vice versa logarithmic) characteristic is suitable for voice transmission, but, as stated in [3], it can seriously reduce the data transfer speed of a modem that is designed to work on an analog channel.

The main disadvantage of the prototype system is the presence of non-linearity in it - a compressor on the transmitting side and an expander - on the receiving side, which do not allow modems that transmit data via the analog channel to work normally. Their direct exclusion causes an increase in the bit depth of the code word of the CA from 8 to 13, which increases the transmission speed and frequency band of the digital stream to an unacceptable value.

The technical result of PM is to reduce the transmission speed and narrow the frequency band of the digital stream with the exception of the compander, which is carried out using the introduced elements and changing the connections in the ADC and DAC.

The essence of PM consists in the fact that in a digital communication system consisting of a source of an analog speech signal (PC), a pulse generator, a time delay unit and series-connected signal sampler, a half-wave signal rectifier, AIM-1 to AIM modulation converter 2, an encoder, a parallel-to-serial code converter, a signal amplifier, wherein the first input of the code converter is connected to the output of the sampler, and the pulse generator is connected to the second input of the sampler directly and to the second input of the AIM-1 to AIM-2 converter through a time delay unit, and on the receiving side, from a series-connected amplifier, pulse regenerator, serial code to digital signal converter (DS) to parallel, decoder, signal multiplier, lower filter frequencies, as well as from the recipient of information, moreover, the second input of the signal multiplier is connected to the first output of the code converter, the clock input of which is connected to the second input of the pulse regenerator, additionally introduced on the transmitting side, a 2-fold divider of the PC frequency band, through which the PC source is connected to the low-frequency input of the sampler, and on the receiving side, a 2-fold frequency band doubler, through which the receiver of information is connected to the low-pass filter, and the resolution of the encoder and decoder increases from 8 to 13, and code converters from 8 to 14.

A significant difference between the PM are the introduced divider and multiplier of the frequency band in half and the communication in the ADC and DAC, since only they allow you to achieve the specified technical result.

PM is illustrated in the drawing.

Figure 1 presents the structural diagram of a single-channel CA, where indicated:

1 - source of an analog speech signal (PC);

2 - PC bandwidth divider 2 times;

3 - time discretizer (AIM-1);

4 - pulse generator;

5 - a half-wave rectifier;

6 - converter AIM-1 to AIM-2;

7 - block time delay pulses;

8 - encoder;

9 - a parallel to serial code converter;

10 - transmitter amplifier;

11 - communication line;

12 - receiver amplifier;

13 - pulse regenerator;

14 - serial to parallel converter;

15 - decoder;

16 - signal multiplier;

17 - low-pass filter (low-pass filter);

18 - PC frequency doubler;

19 - the recipient of information.

Entered elements are surrounded by dashed lines.

The operation of the circuit is as follows.

On the transmitting side, the speech signal (PC) b (t) = U (t) cosφ (t) from block 1 is fed to the input of block 2, where the frequency band of this signal is divided by 2 times. From the output of block 2, the signal u ₂ (t) = U (t) cos0.5φ (t) is fed to the low-frequency input of sampler 3, the high-frequency input of which is supplied from the generator 4 pulses of short duration, the repetition rate of which corresponds to Kotelnikov's theorem and is equal to 4 kHz . From the output of block 3, bipolar readings are converted to unipolar in a half-wave rectifier 5, and then they are fed to the input of block 6, to the second input of which pulses from generator 4, delayed in block 7 for a duration of τ, are supplied. In block 6, samples of short duration are converted into rectangular samples of duration m without changing their amplitude, i.e. in block 6, the AIM-1 modulation of block 3 is converted to AIM-2 modulation. From the output of block 6, the PC samples are input to the encoder 8, which converts their amplitude into a digital signal with a parallel code bit length of n = 13. The digital signal of the parallel code from block 8 is fed to the inputs of block 9, starting from the second and the 14th. The first input of block 9 is fed samples from the sampler 3 to record their sign. Block 9 converts the 14-bit parallel code of the CA to serial, after which the CA is amplified in block 10 and enters the communication line 11.

On the receiving side, the signal from the communication line 11 enters through the amplifier 12 of the receiver to the input of the pulse regenerator 13, and from its output to the information input of the serial to parallel converter 14. The clock input of block 14 (shift register) receives pulses from block 13. The outputs of block 14, starting from the second and the 14th, are connected to the inputs of the decoder 15, which converts the digital signal into consecutive samples of the speech signal of one sign (module), arriving at one the input of the multiplier 16. A pulse of the corresponding sign for a given sample from the first output of block 14 is supplied to the dredge its input; as a result, the output of the multiplier 16 restores the reference sign received at the input of the low-pass filter 17. At the output of this low-pass filter, an analog PC with times 2 times divided frequency band. To restore the frequency band, the signal from the low-pass filter 17 is fed to the input of the multiplier 18 of the frequency band 2 times, after which the undistorted received signal is received by the information receiver 19.

The technical and economic effect of PM is to provide the possibility of transmitting data through an analog communication channel by eliminating a non-linear block - a compander, which is possible if there is a divider introduced in 2 times the frequency band on the transmitting side and a multiplier twice the frequency band on the receiving side when the codeword is bit 14, not 8. Indeed, in the presence of a compander and n = 8, the DS bandwidth Δf _c = Δf _a · n = 3.1 · 8 = 24.8 kHz, and the transmission speed R = 8 · 8 = 64 kbit / from. And in the absence of a compander and in the presence of a frequency band divider twice Δf _c = 0.5 · Δf _a · 14 = 21.72 <4.8 kHz and R = 4 · 14 = 56 <64 kbit / s.

Claims (1)

A digital communication system consisting of a source of an analog speech signal (PC), a pulse generator, a time delay unit and series-connected signal sampler, a half-wave signal rectifier, an AIM-1 to AIM-2 modulation converter, an encoder, a parallel code converter in serial, signal amplifier, and the first input of the code converter is connected to the output of the sampler, and the pulse generator is connected to the second input of the sampler directly and to the second input to the AIM-1 to AIM-2 converter through a time delay unit, and on the receiving side, from a series-connected amplifier, pulse regenerator, serial code to digital signal converter (DS) to parallel, a decoder, a signal multiplier, a low-pass filter, and from the recipient of information, the second input of the signal multiplier connected to the first output of the code converter, the clock input of which is connected to the second output of the pulse regenerator, characterized in that it is additionally introduced into On the receiving side, a 2-fold divider of the PC frequency band, through which the PC source is connected to the low-frequency input of the sampler, and on the receiving side, a 2-fold frequency band doubler, through which the receiver of information is connected to the low-pass filter, and the bit width of the encoder and decoder increases from 8 to 13 , and code converters from 8 to 14.