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Standard frequency band in telephony. What is Frequency Modulation Bandwidth, Spectrum and Sidebands? Modems for dedicated telephone lines

Stations are divided into analog and digital based on the type of switching. Telephone communication, which operates on the basis of converting speech (voice) into an analog electrical signal and transmitting it over a switched communication channel (analog telephony), has long been the only means of transmitting voice messages over a distance. The ability to sample (by time) and quantize (by level) the parameters of an analog electrical signal (amplitude, frequency or phase) made it possible to convert an analog signal into a digital (discrete) one, process it using software methods and transmit it over digital telecommunication networks.

To transmit an analog voice signal between two subscribers in the PSTN (public telephone network) network, a so-called standard voice frequency (VoF) channel is provided, the bandwidth of which is 3100 Hz. In a digital telephony system, the operations of sampling (in time), quantization (in level), encoding and eliminating redundancy (compression) are performed on an analog electrical signal, after which the data stream thus generated is sent to the receiving subscriber and upon “arrival” at the destination is subjected to reverse procedures.

The speech signal is converted using the appropriate protocol, depending on the network through which it is transmitted. Currently, the most efficient transmission of any discrete (digital) signals, including those carrying speech (voice), is provided by digital electrical networks, which implement packet technologies: IP (Internet Protocol), ATM (Asynchronous Transfer Mode) or FR (Frame Relay).

The concept of digital voice transmission is said to have originated in 1993 at the University of Illinois (USA). During the next flight of the Endeavor shuttle in April 1994, NASA transmitted its image and sound to Earth using computer program. The received signal was sent to the Internet, and anyone could hear the voices of the astronauts. In February 1995, the Israeli company VocalTec offered the first version of the Internet Phone program, designed for owners of multimedia PCs running Windows. Then a private network of Internet Phone servers was created. And thousands of people have already downloaded Internet program Phone from the VocalTec home page and started chatting.

Naturally, other companies very quickly appreciated the prospects offered by the ability to talk while in different hemispheres and without paying for it international calls. Such prospects could not go unnoticed, and already in 1995, a flood of products designed for voice transmission over the Network hit the market.

Today, there are several standardized methods of transmitting information that are most widespread in the digital telephony services market: these are ISDN, VoIP, DECT, GSM and some others. Let's try to briefly talk about the features of each of them.

So what is ISDN?

The abbreviation ISDN stands for Integrated Services Digital Network - a digital network with integration of services. This is the modern generation of the worldwide telephone network, which has the ability to transfer any type of information, including fast and correct data transmission (including voice) High Quality from user to user.

Main advantage ISDN networks is that you can connect several digital or analog devices (telephone, modem, fax, etc.) to one network end, and each can have its own landline number.

A regular telephone is connected to a telephone exchange using a pair of conductors. In this case, only one can be carried out per pair. phone conversation. At the same time, noise, interference, radio, and extraneous voices can be heard in the handset - the disadvantages of analog telephone communication, which “collects” all the obstacles in its path. When using ISDN, a network termination is installed for the subscriber, and the sound, converted by a special decoder into a digital format, is transmitted through a specially designated (also completely digital) channel to the receiving subscriber, while ensuring maximum audibility without interference and distortion.

The basis of ISDN is a network built on the basis of digital telephone channels (also providing the possibility of packet-switched data transmission) with a data transfer rate of 64 kbit/s. ISDN services are based on two standards:

Basic access (Basic Rate Interface (BRI)) - two B-channels 64 kbit/s and one D-channel 16 kbit/s

Primary access (Primary Rate Interface (PRI)) - 30 B-channels 64 kbps and one D-channel 64 kbps

Typically, BRI bandwidth is 144 Kbps. When working with PRI, the entire digital communications backbone (DS1) is fully utilized, resulting in throughput 2 Mbit/s. The high speeds offered by ISDN make it ideal for a wide range of modern communications services, including high-speed data transfer, screen sharing, video conferencing, large file transfer for multimedia, desktop video telephony and Internet access.

Strictly speaking, ISDN technology is nothing more than one of the varieties of “computer telephony”, or, as it is also called CTI telephony (Computer Telephony Integration).

One of the reasons for the emergence of CTI solutions was the emergence of requirements to provide company employees with additional telephone services that were either not supported by the existing corporate telephone exchange, or the cost of purchasing and implementing a solution from the manufacturer of this exchange was not comparable with the convenience achieved.

The first signs of CTI service applications were systems of electronic secretaries (autoattended) and automatic interactive voice greetings (menus), corporate voice mail, answering machines and conversation recording systems. To add the service of a particular CTI application, a computer was connected to the company’s existing telephone exchange. A specialized board was installed in it (first on the ISA bus, then on PCI bus), which connected to the telephone exchange via a standard telephone interface. Software computer running under a specific operating system(MS Windows, Linux or Unix), interacted with the telephone exchange through the program interface (API) of a specialized board and thereby ensured the implementation of an additional service corporate telephony. Almost simultaneously with this, a standard was developed software interface for computer-telephony integration – TAPI (Telephony API)

For traditional telephone systems, CTI integration is carried out as follows: some specialized computer board connected to a telephone exchange and transmits (translates) telephone signals, the state of the telephone line and its changes into “programmed” form: messages, events, variables, constants. The telephone component is transmitted via the telephone network, and the software component is transmitted via a data network or IP network.

What does the integration process in IP telephony look like?

First of all, it should be noted that with the advent of IP telephony, the very perception of a telephone exchange (Private Branch eXchange - PBX) has changed. IP PBX is nothing more than another network service of the IP network, and, like most IP network services, it operates in accordance with the principles of client-server technology, i.e., it assumes the presence of service and client parts. So, for example, the service email in an IP network has a service part - mail server and the client part - the user program (for example Microsoft Outlook). The IP telephony service is structured similarly: the service part - the IP PBX server and the client part - the IP telephone (hardware or software) use a single communication medium - the IP network - to transmit voice.

What does this give the user?

The advantages of IP telephony are obvious. Among them are rich functionality, the ability to significantly improve employee interaction and at the same time simplify system maintenance.

In addition, IP communications are evolving in an open manner due to protocol standardization and global IP penetration. Thanks to the principle of openness in the IP telephony system, it is possible to expand the services provided and integrate with existing and planned services.

IP telephony allows you to build a single centralized management system for all subsystems with differentiated access rights and operate subsystems in regional divisions using local personnel.

The modularity of the IP communications system, its openness, integration and independence of components (unlike traditional telephony) provide additional opportunities for building truly fault-tolerant systems, as well as systems with a distributed territorial structure.

Wireless systems DECT communications:

The wireless access standard DECT (Digital Enhanced Cordless Telecommunications) is the most popular system mobile communications V corporate network, the cheapest and easiest option to install. It allows you to organize wireless communication throughout the entire territory of the enterprise, which is so necessary for “mobile” users (for example, enterprise security or heads of workshops and departments).

The main advantage of DECT systems is that with the purchase of such a phone you get a mini-PBX for several internal numbers almost free of charge. The fact is that you can purchase additional handsets for the DECT base once purchased, each of which receives its own internal number. From any handset you can easily call other handsets connected to the same base, transfer incoming and internal calls, and even carry out a kind of “roaming” - register your handset on another base. The reception radius of this type of communication is 50 meters indoors and 300 meters outdoors.

To organize mobile communications in public networks, networks are used cellular communications GSM and CDMA standards, the territorial effectiveness of which is practically unlimited. These are the standards of the second and third generation of cellular communications, respectively. What are the differences?

Every minute from any base station cellular network several telephones located in its vicinity are trying to contact at once. Therefore, stations must provide “multiple access”, that is, simultaneous operation of several telephones without mutual interference.

In first generation cellular systems (standards NMT, AMPS, N-AMPS, etc.), multiple access is implemented by the frequency method - FDMA (Frequency Division Multiple Access): the base station has several receivers and transmitters, each of which operates at its own frequency, and the radiotelephone tunes to any frequency used in the cellular system. Having contacted the base station on a special service channel, the phone receives an indication of which frequencies it can occupy and tunes to them. This is no different from the way a particular radio wave is tuned.

However, the number of channels that can be allocated at the base station is not very large, especially since neighboring cellular network stations must have different sets of frequencies so as not to create mutual interference. Most second-generation cellular networks began to use the time-frequency method of channel division - TDMA (Time Division Multiple Access). In such systems (and these are networks of GSM, D-AMPS, etc. standards) different frequencies are also used, but each such channel is allocated to the phone not for the entire communication time, but only for short periods of time. The remaining same intervals are alternately used by other phones. Useful information in such systems (including speech signals) is transmitted in “compressed” form and in digital form.

Sharing each frequency channel with several phones makes it possible to provide service to a larger number of subscribers, but there are still not enough frequencies. CDMA technology, built on the principle of code division of signals, was able to significantly improve this situation.

The essence of the code division method used in CDMA is that all phones and base stations simultaneously use the same (and at the same time the entire) frequency range allocated for the cellular network. In order for these broadband signals to be distinguished from each other, each of them has a specific code “coloring” that ensures that it stands out from the others.

Over the past five years, CDMA technology has been tested, standardized, licensed and launched by most wireless equipment vendors and is already in use around the world. Unlike other methods of subscriber access to the network, where signal energy is concentrated on selected frequencies or time intervals, CDMA signals are distributed in a continuous time-frequency space. In fact, this method manipulates frequency, time, and energy.

The question arises: can CDMA systems, with such capabilities, “peacefully” coexist with AMPS/D-AMPS and GSM networks?

It turns out they can. Russian regulatory authorities have allowed the operation of CDMA networks in the radio frequency band 828 - 831 MHz (signal reception) and 873-876 MHz (signal transmission), where two CDMA radio channels with a width of 1.23 MHz are located. In turn, the GSM standard in Russia is allocated frequencies above 900 MHz, so the operating ranges of CDMA and GSM networks do not overlap in any way.

What I want to say in conclusion:

As practice shows, modern users are increasingly gravitating towards broadband services (video conferencing, high-speed data transfer) and increasingly prefer mobile terminal regular wired. If we also take into account the fact that the number of such applicants in large companies can easily exceed a thousand, we get a set of requirements that only a powerful modern digital exchange (PBX) can satisfy.

Today, the market offers many solutions from various manufacturers that have the capabilities of both traditional PBXs, switches or routers for data networks (including ISDN and VoIP technologies), and the properties of wireless base stations.

Digital PBXs today, to a greater extent than other systems, meet the specified criteria: they have the capabilities of switching broadband channels, packet switching, and are simply integrated with computer systems(CTI) and allow the organization of wireless microcells within corporations (DECT).

Which of the following types of communication is better? Decide for yourself.

Almost all electrical signals that display real messages contain an infinite spectrum of frequencies. For undistorted transmission of such signals, a channel with infinite bandwidth would be required. On the other hand, the loss of at least one spectrum component during reception leads to distortion of the temporal shape of the signal. Therefore, the task is to transmit a signal in a limited channel bandwidth in such a way that the signal distortion meets the requirements and quality of information transmission. Thus, the frequency band is a limited (based on technical and economic considerations and requirements for transmission quality) signal spectrum.

The frequency bandwidth ΔF is determined by the difference between the upper FB and lower FH frequencies in the message spectrum, taking into account its limitations. Thus, for a periodic sequence of rectangular pulses, the signal bandwidth can be approximately found from the expression:

where tn is the pulse duration.

The primary telephone signal (voice message), also called subscriber signal, is a non-stationary random process with a frequency band from 80 to 12,000 Hz. Speech intelligibility is determined by formants (amplified regions of the frequency spectrum), most of which are located in the band 300 ... 3400 Hz. Therefore, on the recommendation of the International Advisory Committee on Telephony and Telegraphy (ICITT), an efficiently transmitted frequency band of 300 ... 3400 Hz was adopted for telephone transmission. This signal is called a voice frequency (VF) signal. At the same time, the quality of the transmitted signals is quite high - syllable intelligibility is about 90%, and phrase intelligibility is 99%.

Audio broadcast signals. Sound sources when transmitting broadcast programs are musical instruments or human voice. Range sound signal occupies a frequency band of 20...20000 Hz.

For sufficiently high quality (first class broadcast channels) the ∆FC frequency band should be 50...10000 Hz, for flawless reproduction of broadcast programs (highest class channels) - 30...15000 Hz, second class - 100...6800 Hz.

In broadcast television, the method adopted is to alternately convert each element of the image into an electrical signal and then transmit this signal over one communication channel. To implement this principle, special cathode ray tubes are used on the transmitting side, converting the optical image of the transmitted object into an electrical video signal unfolded in time.

Figure 2.2.1 - Design of the transmitting tube

As an example, Figure 2.2.1 shows a simplified version of one of the transmitting tube options. Inside the glass flask, which is under high vacuum, there is a translucent photocathode (target) and an electronic spotlight (EP). A deflection system (OS) is placed on the outside of the tube neck. The spotlight generates a thin electron beam, which, under the influence of an accelerating field, is directed towards the target. Using a deflection system, the beam moves from left to right (along the lines) and from top to bottom (along the frame), running around the entire surface of the target. The collection of all (N) rows is called a raster. An image is projected onto the tube target, coated with a photosensitive layer. As a result, each elementary section of the target acquires electric charge. A so-called potential relief is formed. The electron beam, interacting with each section (point) of the potential relief, seems to erase (neutralize) its potential. The current that flows through the load resistance Rн will depend on the illumination of the target area where the electron beam hits, and the video signal Uc will be released at the load (Figure 2.2.2). The video signal voltage will vary from a “black” level, corresponding to the darkest areas of the transmitted image, to a “white” level, corresponding to the lightest areas of the image.

2.1.1. Analogue telephone networks

Analog telephone networks refer to wide-area circuit-switched networks that were created to provide public telephone services to the public. Analog telephone networks are focused on a connection that is established before conversations (voice transmission) begin between subscribers. The telephone network is formed (switched) using automatic telephone exchange switches.

Telephone networks consist of:

automatic telephone exchanges (ATS);
telephone sets;
trunk communication lines (communication lines between automatic telephone exchanges);
subscriber lines (lines connecting telephone sets to PBX).

The subscriber has a dedicated line that connects his telephone set to the PBX. Trunk communication lines are used by subscribers in turn.

Analog telephone networks are also used for data transmission as:

access networks to packet-switched networks, for example, Internet connections (both dial-up and leased telephone lines are used);
trunks of packet networks (mainly dedicated telephone lines are used).

The analog circuit-switched telephone network provides services for the packet network physical level, which after switching is a point-to-point physical channel.

Regular telephone network or POTS(Plain Old Telephone Service - old “flat” telephone service) ensures the transmission of a voice signal between subscribers with a frequency range of up to 3.1 kHz, which is quite sufficient for a normal conversation. To communicate with subscribers, a two-wire line is used, through which the signals of both subscribers travel simultaneously in opposite directions during a conversation.

The telephone network consists of many stations that have hierarchical connections among themselves. The switches of these stations pave the way between the calling and called subscriber's telephone exchanges under the control of information provided by the signaling system. Trunk communication lines between telephone exchanges must provide the ability to simultaneously transmit a large amount of information (support a large number of connections).

It is impractical to allocate a separate trunk line for each connection, and for more efficient use of physical lines the following is used:

frequency division multiplexing method;
digital channels and multiplexing of digital streams from multiple subscribers.

Frequency Division Multiplexing (FDM) method

In this case, a single cable transmits multiple channels in which a low-frequency voice signal modulates a high-frequency oscillator signal. Each channel has its own oscillator, and the frequencies of these oscillators are separated enough from each other to transmit signals in a bandwidth of up to 3.1 kHz with a normal level of separation from each other.

Application of digital channels for trunk transmissions

To do this, the analog signal from the subscriber line at the telephone exchange is digitized and then digitally delivered to the recipient's telephone exchange. There it is converted back and transmitted to the analog subscriber line.

To ensure two-way communication at the telephone exchange, each end of the subscriber line has a pair of converters - ADC (analog-to-digital) and DAC (digital-to-analog). For voice communications with a standard bandwidth (3.1 kHz), the quantization frequency is 8 kHz. Acceptable dynamic range (the ratio of the maximum signal to the minimum) is provided with 8-bit conversion.

In total, it turns out that each telephone channel requires a data transfer rate of 64 kbit/s (8 bits x 8 kHz).

Often, signal transmission is limited to 7-bit samples, and the eighth (LSB) bit is used for signaling purposes. In this case, the pure voice stream is reduced to 56 kbit/s.

To effectively use trunk lines, digital streams from multiple subscribers at telephone exchanges are multiplexed into channels of various capacities that connect telephone exchanges to each other. At the other end of the channel, demultiplexing is performed - separating the required stream from the channel.

Multiplexing and demultiplexing, of course, is carried out at both ends simultaneously, since telephone communication is two-way. Multiplexing is carried out using time division (TDM – Time Division Multiplexing).

In a backbone channel, information is organized as a continuous sequence of frames. Each subscriber channel in each frame is allocated the time interval during which data from this channel is transmitted.

Thus, in modern analog telephone lines, analog signals are transmitted over the subscriber line, and digital signals are transmitted in trunk lines.

Modems for dial-up analog telephone lines

Public telephone networks, in addition to voice transmission, allow the transmission of digital data using modems.

A modem (modulator-demodulator) is used to transmit data over long distances using dedicated and switched telephone lines.

The modulator converts binary information coming from the computer into analog signals with frequency or phase modulation, the spectrum of which corresponds to the bandwidth of ordinary voice telephone lines. The demodulator extracts the encoded binary information from this signal and transmits it to the receiving computer.

Fax modem (fax-modem) allows you to send and receive fax images, compatible with conventional fax machines.

Modems for dedicated telephone lines

Leased physical lines have a much wider bandwidth than switched lines. Special modems are produced for them, providing data transmission at speeds of up to 2048 kbit/s and over considerable distances.

xDSL technologies

xDSL technologies are based on converting the subscriber line of a regular telephone network from analog to digital xDSL (Digital Subscriber Line). The essence of this technology is that splitter filters are installed at both ends of the subscriber line - at the telephone exchange and at the subscriber's.

The low-frequency (up to 3.5 kHz) component of the signal is fed to conventional telephone equipment (PBX port and telephone set at the subscriber), and the high-frequency (above 4 kHz) is used for data transmission using xDSL modems.

xDSL technologies allow you to simultaneously use the same telephone line for both data transmission and voice transmission (telephone conversations), which is not possible with conventional dial-up modems.

Ensuring the transmission of electrical communication signals in an effectively transmitted frequency band (ETF) of 0.3 - 3.4 kHz. In telephony and communications, the abbreviation KTC is often used. The audio channel is a unit of measurement for the capacitance (density) of analog transmission systems (eg K-24, K-60, K-120). At the same time for digital systems transmission (for example, PCM-30, PCM-480, PCM-1920) the unit of measurement of capacitance is the main digital channel.

Efficiently transmitted frequency band- frequency band, the residual attenuation at extreme frequencies of which differs from the residual attenuation at a frequency of 800 Hz by no more than 1 Np at the maximum communication range characteristic of a given system.

The width of the EPCH determines the quality of telephone transmission, and the possibility of using the telephone channel to transmit other types of communications. In accordance with the international standard for telephone channels of multi-channel equipment, the frequency range is set from 300 to 3400 Hz. With such a band, a high degree of speech intelligibility is ensured, its sound is well natural, and great opportunities are created for secondary multiplexing of telephone channels.

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Subtitles

PM channel operating modes

Purpose of modes

2 PR. OK - for open telephone communication in the absence of transit extenders on the telephone switch;
2 PR. TR - for temporary transit connections of open telephone channels, as well as for terminal communication if there are transit extenders on the telephone switch;
4 PR OK - for use in networks of multi-channel voice-frequency telegraph, closed telephone communication, data transmission, etc., as well as for transit connections with significant lengths of connecting lines;
4 PR TR - for long-term transit connections.

Almost all electrical signals that display real messages contain an infinite spectrum of frequencies. For undistorted transmission of such signals, a channel with infinite bandwidth would be required. On the other hand, the loss of at least one spectrum component during reception leads to distortion of the temporal shape of the signal. Therefore, the task is to transmit a signal in a limited channel bandwidth in such a way that the signal distortion meets the requirements and quality of information transmission. Thus, a frequency band is a limited (based on technical and economic considerations and requirements for transmission quality) signal spectrum.

The frequency bandwidth ΔF is determined by the difference between the upper F B and lower F H frequencies in the message spectrum, taking into account its limitations. Thus, for a periodic sequence of rectangular pulses, the signal bandwidth can be approximately found from the expression:

where t n is the pulse duration.

1.Primary telephone signal (voice message), also called subscriber, is a non-stationary random process with a frequency band from 80 to 12,000 Hz. Speech intelligibility is determined by formants (amplified regions of the frequency spectrum), most of which are located in the band 300 ... 3400 Hz. Therefore, on the recommendation of the International Advisory Committee on Telephony and Telegraphy (ICITT), an efficiently transmitted frequency band of 300 ... 3400 Hz was adopted for telephone transmission. This signal is called a voice frequency (VF) signal. At the same time, the quality of the transmitted signals is quite high - syllable intelligibility is about 90%, and phrase intelligibility is 99%.

2.Audio broadcast signals . Sound sources when transmitting broadcast programs are musical instruments or human voice. The spectrum of the audio signal occupies the frequency band 20...20000 Hz.

For sufficiently high quality (first class broadcast channels) the frequency band ∆F C should be 50...10000 Hz, for flawless reproduction of broadcast programs (highest class channels) - 30...15000 Hz, second class - 100...6800 Hz.

3. In broadcast television a method has been adopted for sequentially converting each image element into an electrical signal and then transmitting this signal through one communication channel. To implement this principle, special cathode ray tubes are used on the transmitting side, converting the optical image of the transmitted object into an electrical video signal unfolded in time.

Figure 2.6 – Design of the transmitting tube

As an example, Figure 2.6 shows a simplified version of one of the transmitting tube options. Inside the glass flask, which is under high vacuum, there is a translucent photocathode (target) and an electronic spotlight (EP). A deflection system (OS) is placed on the outside of the tube neck. The spotlight generates a thin electron beam, which, under the influence of an accelerating field, is directed towards the target. Using a deflection system, the beam moves from left to right (along the lines) and from top to bottom (along the frame), running around the entire surface of the target. The collection of all (N) rows is called a raster. An image is projected onto the tube target, coated with a photosensitive layer. As a result, each elementary section of the target acquires an electric charge. A so-called potential relief is formed. The electron beam, interacting with each section (point) of the potential relief, seems to erase (neutralize) its potential. The current that flows through the load resistance R n will depend on the illumination of the target area that the electron beam hits, and a video signal U c will be released at the load (Figure 2.7). The video signal voltage will vary from a “black” level, corresponding to the darkest areas of the transmitted image, to a “white” level, corresponding to the lightest areas of the image.

Figure 2.7 – The shape of a television signal in a time interval where there are no frame pulses.

If the “white” level corresponds to the minimum signal value, and the “black” level corresponds to the maximum, then the video signal will be negative (negative polarity). The nature of the video signal depends on the design and operating principle of the transmitting tube.

The television signal is a pulsed unipolar (since it is a function of brightness, which cannot be multipolar) signal. It has a complex shape and can be represented as the sum of constant and harmonic components of oscillations of various frequencies.
The DC component level characterizes the average brightness of the transmitted image. When transmitting moving images, the value of the constant component will continuously change in accordance with the illumination. These changes are happening very quickly low frequencies(0-3 Hz). Using the lower frequencies of the video signal spectrum, large image details are reproduced.

Television, as well as light cinema, became possible thanks to the inertia of vision. The nerve endings of the retina continue to be excited for some time after the cessation of the light stimulus. At a frame rate F k ≥ 50 Hz, the eye does not notice the intermittency of the image change. In television, the time for reading all N lines (frame time - Tk) is chosen equal to Tk = s. To reduce image flickering, interlaced scanning is used. First, in a half-frame time equal to T p/c = s, all odd lines are read one by one, then, in the same time, all even lines are read. The frequency spectrum of the video signal will be obtained when transmitting an image that is a combination of the light and dark half of the raster (Figure 2.8). The signal represents pulses close in shape to rectangular. The minimum frequency of this signal during interlaced scanning is the frequency of the fields, i.e.

Figure 2.8 – To determine the minimum frequency of the television signal spectrum

With the help of high frequencies, the finest details of the image are transmitted. Such an image can be represented in the form of small black and white squares alternating in brightness with sides equal to the diameter of the beam (Figure 2.9, a), located along the line. Such an image will contain the maximum number of image elements.

Figure 2.9 – To determine the maximum frequency of the video signal

The standard provides for the decomposition of an image in a frame into N = 625 lines. The time to draw one line (Fig. 2.9, b) will be equal to . A signal that changes along the line is obtained when black and white squares alternate. The minimum signal period will be equal to the time it takes to read a pair of squares:

where n pairs is the number of pairs of squares in a line.

The number of squares (n) in the line will be equal to:

where is the frame format (see Figure 2.2.4, a),

b – width, h – height of the frame field.

Then ; (2.10)

The frame format is assumed to be k=4/3. Then the upper frequency of the signal F in will be equal to:

When transmitting 25 frames per second with 625 lines each, the nominal line frequency (line frequency) is 15.625 kHz. The upper frequency of the television signal will be 6.5 MHz.

According to the standard adopted in our country, the voltage of the complete video signal U TV, consisting of synchronization pulses U C, a brightness signal and damping pulses U P, is U TV = U P + U C = 1V. In this case, U C = 0.3 U TV, and U P =0.7 U TV. As can be seen from Figure 2.10, the signal soundtrack is located higher in the spectrum (fn 3V = 8 MHz) of the video signal. Typically, a video signal is transmitted using amplitude modulation (AM), and an audio signal using frequency modulation (FM).

Sometimes, in order to save channel bandwidth, the upper frequency of the video signal is limited to the value Fv = 6.0 MHz, and the audio carrier is transmitted at a frequency fн з = 6.5 MHz.

Figure 2.10 – Placement of spectra of image and sound signals in a television broadcast radio channel.

Workshop (similar tasks are included in the exam papers)

Task No. 1: Find the pulse repetition rate of the transmitted signal and the signal bandwidth if there are 5 pairs of black and white alternating vertical stripes on the TV screen

Task No. 2: Find the pulse repetition rate of the transmitted signal and the signal bandwidth if there are 10 pairs of black and white alternating horizontal stripes on the TV screen

When solving problem No. 1, it is necessary to use the known duration of one line of a standard TV signal. During this time, there will be a change of 5 pulses corresponding to the black level and 5 pulses corresponding to the white level (you can calculate their duration). In this way, the pulse frequency and signal bandwidth can be determined.

When solving problem No. 2, proceed from the total number of lines in the frame, determine how many lines are in one horizontal stripe, keep in mind that scanning is carried out interlaced. This way you will determine the duration of the pulse corresponding to the black or white level. Continue as in task No. 1

When preparing the final work, for convenience, use graphic image signals and spectra.

4. Fax signals. Facsimile (phototelegraph) communication is the transmission of still images (drawings, drawings, photographs, texts, newspaper strips, and so on). The fax message (image) conversion device converts the light flux reflected from the image into an electrical signal (Figure 2.2.6)

Figure 2.11 - Functional diagram of fax communication

Where 1 is the fax communication channel; 2 – drive, synchronizing and phasing devices; 3 – transmitting drum, on which the original of the transmitted image on paper is placed; FEP – photoelectronic converter of reflected light flux into an electrical signal; OS – optical system for forming a light beam.

When transmitting elements alternating in brightness, the signal takes the form of a pulse sequence. The frequency of repetition of pulses in a sequence is called the pattern frequency. The pattern frequency, Hz, reaches its maximum value when transmitting an image whose elements and the spaces separating them are equal to the dimensions of the scanning beam:

F rismax = 1/(2τ u) (2.12)

where τ u is the pulse duration equal to the transmission duration of the image element, which can be determined through the parameters of the scanning device.

So, if π·D is the length of the line, and S is the scan pitch (the diameter of the scanning beam), then there are π·D/S elements in the line. At N revolutions per minute of a drum having a diameter D, the image element transmission time, measured in seconds:

The minimum frequency of the picture (when changing along the line), Hz, will be when scanning an image containing black and white stripes along the length of the line, equal in width to half the length of the line. Wherein

F pус min = N/60, (2.14)

To perform phototelegraph communication of satisfactory quality, it is enough to transmit frequencies from F pic min to F pic max. The International Telegraph and Telephony Advisory Committee recommends N = 120, 90 and 60 rpm for fax machines; S = 0.15 mm; D = 70 mm. From (2.13) and (2.14) it follows that at N = 120 F rice max = 1466 Hz; F fig min = 2 Hz; at N =60 F fig max = 733 Hz; F fig min = 1 Hz; The dynamic range of the fax signal is 25 dB.

Telegraph and data signals. Messages and signals of telegraphy and data transmission are discrete.

Devices for converting telegraph messages and data represent each message character (letter, number) in the form of a certain combination of pulses and pauses of the same duration. A pulse corresponds to the presence of current at the output of the conversion device, a pause corresponds to the absence of current.

For data transmission, more complex codes are used, which make it possible to detect and correct errors in the received combination of pulses that arise from interference.

Devices for converting telegraph signals and transmitting data into messages use the received combinations of pulses and pauses to restore message characters in accordance with the code table and output them to a printing device or display screen.

The shorter the duration of the pulses displaying messages, the more of them will be transmitted per unit of time. The reciprocal of the pulse duration is called the telegraphing speed: B = 1/τ and, where τ and is the pulse duration, s. The unit of telegraph speed was called the baud. With a pulse duration of τ and = 1 s, the speed is B = 1 Baud. Telegraphy uses pulses with a duration of 0.02 s, which corresponds to a standard telegraphy speed of 50 baud. Data transfer rates are significantly higher (200, 600, 1200 baud and more).

Telegraphy and data transmission signals usually take the form of sequences of rectangular pulses (Figure 2.4, a).

When transmitting binary signals, it is enough to fix only the sign of the pulse for a bipolar signal, or the presence or absence for a unipolar signal. Pulses can be reliably detected if they are transmitted using a bandwidth that is numerically equal to the baud rate. For a standard telegraph speed of 50 baud, the spectrum width of the telegraph signal will be 50 Hz. At 2400 baud (medium-speed data transmission system), the signal spectrum width is approximately 2400 Hz.

5. Average message power P SR is determined by averaging the measurement results over a long period of time.

The average power that a random signal s(t) develops across a 1 Ohm resistor:

The power contained in a finite frequency band between ω 1 and ω 2 is determined by integrating the function G(ω) β within the corresponding limits:

The function G(ω) represents the spectral density of the average power of the process, that is, the power contained in an infinitesimal frequency band.

For convenience of calculations, power is usually given in relative units, expressed in logarithmic form (decibels, dB). In this case the power level is:

If the reference power R E = 1 mW, then p x is called the absolute level and is expressed in dBm. Taking this into account, the absolute level of average power is:

Peak power p peak (ε %) – this is the message power value that can be exceeded for ε % of the time.

The signal crest factor is determined by the ratio of the peak power to the average message power, dB,

From the last expression, dividing the numerator and denominator by RE, taking into account (2.17) and (2.19), we determine the peak factor as the difference between the absolute levels of peak and average powers:

The dynamic range D (ε%) is understood as the ratio of the peak power to the minimum message power P min . The dynamic range, like the crest factor, is usually estimated in dB:

The average power of the voice frequency signal, measured during busy hours (BHH), taking into account control signals - dialing, calling, etc. - is 32 μW, which corresponds to a level (compared to 1 mW) p av = -15 dBm

Maximum power telephone signal, the probability of exceeding which is negligibly small is equal to 2220 μW (which corresponds to a level of +3.5 dBm); The minimum signal power that can still be heard against the background noise is taken to be 220,000 pW (1 pW = 10 -12 mW), which corresponds to a level of 36.5 dBm.

The average power P CP of the broadcast signal (measured at a point with zero relative level) depends on the averaging interval and is equal to 923 μW when averaged over an hour, 2230 μW per minute and 4500 μW per second. The maximum broadcast signal power is 8000 μW.

The dynamic range of D C broadcast signals is 25...35 dB for announcer speech, 40...50 dB for an instrumental ensemble, and up to 65 dB for a symphony orchestra.

Primary discrete signals are usually in the form of rectangular pulses of direct or alternating current, usually with two resolved states (binary or on-off).

The modulation rate is determined by the number of units (chips) transmitted per unit of time, and is measured in baud:

B = 1/τ u, (2.23)

where τ and is the duration of an elementary message.

The speed of information transmission is determined by the amount of information transmitted per unit of time and is measured in bits/s:

where M is the number of signal positions.

In binary systems (M=2), each element carries 1 bit of information, therefore, according to (2.23) and (2.24):

C max =B, bit/s (2.25)

Control questions

1. Define the concepts “information”, “message”, “signal”.

2. How to determine the amount of information in a single message?

3. What types of signals are there?

4. How does a discrete signal differ from a continuous signal?