Frequency band in telephony. Digital and analogue telephony: differences and prospects. Leased lines, dial-up lines

2.1.1. Analogue telephone networks

Analog telephone networks are classified as global networks with circuit switching, which were created to provide public telephone services to the population. 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 effective use physical lines are 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 digital form 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 communication with a standard bandwidth (3.1 kHz), a quantization frequency of 8 kHz is adopted. 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

Telephone networks common use, in addition to voice transmission, allow you to transmit digital data using modems.

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

Modulator coming from the computer binary information converts into analog signals with frequency or phase modulation, the spectrum of which corresponds to the bandwidth of conventional 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 and voice transmission ( telephone conversations), which conventional dial-up modems do not allow.

Usually we don't care how the telephone line works (but not when we have to shout at the top of our lungs into the phone: “Please repeat, I can’t hear anything!”).

Telephone companies provide a wide variety of services to customers. It is not so easy to understand the price lists of these services - what is actually offered, and how much you should pay for which service. In this article we will not say a word about prices, but we will try to find out what the differences are between the most commonly offered products and services in the field. telephone communication.

ANALOG LINES, DIGITAL LINES

Firstly, lines can be analog and digital. The analog signal changes continuously; it always has a specific value, representing, for example, the volume and pitch of the voice being transmitted, or the color and brightness of a certain area of the image. Digital signals only have discrete values. As a rule, the signal is either on or off, either it is present or it is not. In other words, its value is either 1 or 0.

Analog telephone lines have been used in telephony since time immemorial. Even fifty-year-old telephones can most likely be connected to a local loop - the line between the home telephone socket and the central telephone exchange. (A telephone central office is not a shiny skyscraper in the center of a city; the average subscriber loop is no more than 2.5 miles (four kilometers) long, so a "telephone central office" is usually located in some nondescript building nearby.)

During telephone conversation The microphone built into the handset converts speech into an analog signal, transmitted to the central telephone exchange, from where it goes either to another subscriber loop or to other switching devices if the called number is outside the coverage area of this exchange. When dialing a number, the telephone generates in-band signals transmitted over the same main channel, indicating who the call is intended for.

Over the course of their existence, telephone companies have accumulated extensive experience in voice transmission. It has been established that the frequency range from 300 to 3100 Hz is generally sufficient to perform this task. Let us recall that hi-fi audio systems are capable of reproducing sound without distortion in the frequency range of 20-20000 Hz, which means that the telephone range is usually only enough for the subscriber to recognize the caller by voice (for other applications this range is likely to be too much narrow - for transmitting music, for example, telephone communication is completely unsuitable). Telephone companies provide a smooth decline in the amplitude-frequency response at high and low frequencies using an analog telephone channel of 4000 Hz.

The central telephone exchange, as a rule, digitizes the signal intended for further transmission over the telephone network. With the exception of Gilbeth County (Arkansas) and Rat Fork (Wyoming), all American telephone networks transmit signals between central stations digitally. Although many companies use digital private branch exchanges and data transmission facilities, and all ISDN facilities are based on digital encoding, subscriber loops are still the “last bastion” of analog communications. This is explained by the fact that most telephones in private homes do not have means of signal digitization and cannot work with lines with a bandwidth of over 4000 Hz.

WHAT IS 4000 Hz ENOUGH FOR?

A modem is a device that converts computer digital signals into analog signals with frequencies within the bandwidth of a telephone line. The maximum throughput of a link is directly related to the bandwidth. More precisely, the amount of throughput (in bits/sec) is determined by the bandwidth and the signal-to-noise ratio tolerance. Currently, the maximum throughput of modems - 33.6 Kbps - is already close to this limit. Users of 28.8 Kbps modems are well aware that noisy analog lines rarely provide full bandwidth. throughput, which often turns out to be much lower. Compression, caching and other tricks help to somewhat improve the situation, but nevertheless, we are more likely to live to see the invention of a perpetual motion machine than to see the advent of modems with a bandwidth of 50 or at least 40 Kbps on ordinary analog lines.

Telephone companies solve the opposite problem - digitize the analog signal. To transmit the resulting digital signal, channels with a bandwidth of 64 Kbit/s are used (this is the world standard). This channel, called DS0 (digital signal, level zero), is the basic building block from which all other telephone lines are built. For example, you can combine (the correct term is condense) 24 DS0 channels into a DS1 channel. By renting a T-1 line, the user actually receives a DS1 channel. When calculating the total throughput of DS1, we must remember that after every 192 data bits (that is, 8000 times per second), one synchronization bit is transmitted: a total of 1.544 Mbps (64000 times 24 plus 8000).

LEased lines, switched lines

In addition to the T-1 line, the client can rent leased lines or use regular switching lines. By renting a T-1 circuit or a low-speed data line, such as a dataphone digital service (DDS), from a telephone company, the subscriber is effectively renting a direct connection and, as a result, becomes the sole user of a 1.544 Mbps circuit (T-1 ) or 56 Kbit/s (low-speed line).

Although frame relay technology involves switching individual frames, the corresponding services are offered to the user in the form of virtual communication channels between fixed endpoints. From a network architecture point of view, frame relay should be considered more of a dedicated line than a dial-up line; It is also important that the price of such a service for the same bandwidth is significantly lower.

Switching services (an example of this would be the service of a regular residential telephone) are services purchased from the telephone company. Upon request, the subscriber is provided with a connection via a network of public switches to any node of the telephone network. Unlike the situation with leased lines, in this case the fee is charged for the connection time or the actual volume of traffic and depends largely on the frequency and volume of network use. Digital communications services can be provided based on X.25, Switched 56, ISDN Basic Rate Interface (BRI), ISDN Primary Rate Interface (PRI), Switched Multimegabit Data Service (SMDS) and ATM protocols. Some organizations, such as universities, railways or municipal organizations create private networks using their own switches and leased, and sometimes even their own lines.

If the line received from the telephone company is digital, then for communication between telephone network and terminal equipment (telephone company term for equipment such as computers, fax machines, video phones, and digital telephones) do not need to convert digital signals to analog, and therefore eliminate the need for a modem. However, in this case, the use of the telephone network imposes certain requirements on the subscriber. In particular, you should ensure that the local loop is terminated correctly, that traffic is carried correctly, and that the telephone company's diagnostics are supported.

The line supporting the ISDN BRI protocol must be connected to a device called NT1 (network termination 1). In addition to terminating the line and supporting diagnostic procedures, the NT1 device coordinates a two-wire subscriber loop with a four-wire digital terminal equipment system. When using T-1 or DDS leased digital lines or digital communications services, a channel service unit (CSU) should be used as line load. The CSU works as a terminator, ensures correct line load and processes diagnostic commands. The client's terminal equipment interacts with a data service unit (DSU), which converts digital signals to a standard form and transmits them to the CSU. Structurally, CSU and DSU are often combined into one module called CSU/DSU. The DSU can be built into a router or multiplexer. Thus, in this case (although modems are not needed here), the installation of certain interface devices will be required.

MEDIA FOR TELEPHONE COMMUNICATIONS

Most analog subscriber loops can provide a throughput of 33.6 Kbps only under very favorable conditions. On the other hand, the same twisted pair, connecting the office to the central telephone exchange, can easily be used to work with ISDN BRI, which gives a data throughput of 128 Kbps and another 16 Kbps for management and configuration. What's the matter? The signal transmitted over analog telephone lines is filtered to suppress all frequencies above 4 kHz. When using digital lines, such filtering is not required, so the bandwidth of the twisted pair is significantly wider, and consequently, the throughput increases.

Leased lines with a capacity of 56 and 64 Kbps are two-wire or four-wire digital lines (in the latter case, one pair is used for transmission and the other for reception). The same lines are suitable as a carrier for providing digital communication services, for example, frame relay or Switched 56. Four-wire lines or even optical cables are often used as a carrier for T-1, as well as ISDN PRI and frame relay. T-3 lines are sometimes coaxial cable, but more often they are still performed on the basis of optical.

Although ISDN continues to attract widespread attention as a means of high-speed signal transmission over long distances, Lately Newer means of communication have appeared for the “last mile” (i.e., subscriber loop). PairGain and AT&T Paradyne offer products based on Bellcore's high-bit-rate digital subscriber loop (HDSL) technology. These products allow you to equalize the capabilities of all existing subscriber loops; By installing HDSL devices at both ends of the line, you can get DS1 throughput (1.544 Mbit/s) on almost all existing subscriber loops. (HDSL up to 3.7 km long can be used on subscriber loops without repeaters in the case of standard 24-gauge wires. For regular T-1 lines to work, repeaters must be installed every kilometer and a half). An alternative to HDSL in achieving last-mile DS1 throughput is to either use optical cable (which is quite expensive) or install multiple repeaters on each line (not as expensive as fiber optic technology, but still not cheap). In addition, in this case, the costs of the telephone company, and therefore the client, to maintain the line in working order increase significantly.

But even HDSL is not the last word in technology in the field of increasing capacity in the last mile. HDSL's successor, asymmetrical digital subscriber line (ASDL) technology, is expected to provide 6 Mbps throughput in one direction; the other's throughput is significantly lower - something around 64 Kbps. Ideally, or at least in the absence of a monopoly - assuming that the cost of the service to the customer is approximately equal to its cost to the telephone company - a large share of customers could use ISDN PRI (or other T-1 based services) at a cost , comparable to the current ISDN BRI price.

Today, however, ISDN proponents likely have little to worry about; In most cases, telephone companies will choose to increase line capacity and pocket all the profits without reducing the cost of service to the customer. It is not at all obvious that tariffs for services should be based on common sense.

Table 1. Types of telephone services

Line type	Service	Type of switching	Subscriber loop carrier
Analogue line		Line switching	Two-wire twisted pair
DS0(64 Kbps)	DDS (leased line)	Leased line
		PVC with switching	Two- or four-wire twisted pair
		Switching	Two- or four-wire twisted pair
		Line switching	Two- or four-wire twisted pair
		Line switching	Two- or four-wire twisted pair
		Line switching	Two-wire twisted pair
Multiple DS0 (from 64 Kbps to 1536 Mbps 64 Kbps increments)		Leased line	Two- or four-wire twisted pair
		PVC with switching	Two- or four-wire twisted pair
(1544 Mbps) (24 DS0 lines)	Leased line T-1	Leased line
		PVC with switching	Four-wire twisted pair or fiber optic
		Packet switching	Four-wire twisted pair or fiber optic
		Line switching	Four-wire twisted pair or fiber optic
(44736 Mbps) (28 DS1 lines, 672 DS0 lines)		Cellular switching
(44736 Mbps) (28 DS1 lines, 672 DS0 lines)		Packet switching	Coaxial cable or fiber optic

Steve Steinke can be reached via the Internet at:

When transmitting a signal over long distances, it is energetically beneficial to use high-frequency carrier parameters of which modulated transmitted signal. For transmission

voices over communication channels typically use two carrier modulation methods: amplitude(AM) and frequency(World Cup). However, fixed-line systems used only AM, for which the required channel bandwidth transmission was 2, where - stripe frequencies occupied by the signal (namely, the CFC). Moreover, using compaction technology based on AM-OBP, it was possible to filter out the left or right side components, as well as suppress the carrier (CA) and form the desired channel path (see Chapter 3, Fig. 3-3).

We associate communication systems with voice or telephone communication systems, which only in the last 20 years (in connection with the development of modem communications) began to be used for data transmission. These systems were designed and optimized for speech transmission and were built as multichannel, using various methods channel seals for cable transmission of more and more channels.

Considering that the CFC frequency band (300-3400 Hz) had to be filtered by a real, and not an ideal, analog band-pass filter, it was proposed to use a 4 kHz band as the design baseband width standard telephone channel(STK), protective strip between two adjacent channels at the same time it was 900 Hz, which made it possible to significantly reduce the transient interference between telephone channels.

Pulse code modulation (PCM)

Digital signal representation methods were based on the process sampling transmitted voice signal, i.e. use samples, taken periodically from sampling frequency f d. It was selected from the condition of subsequent (at the reception point) restoration of the signal without loss using a low-pass filter (with f cp=4 kHz) based Kotelnikov-Nyquist theorem, claiming that spectrum-limited signalf cpM.B. restored without loss if the sampling frequency is at least f d=2 f cp. From this we found that for STC the sampling frequency is 8 kHz, i.e. samples should be taken from sampling period T d = 125 µs, (Fig. 8-1).

Fig.8-1. Convert analog signal to digital PCM signal

Next step - quantization sample amplitudes, i.e. definition of its equivalent digital value. The specified steps to be taken when pulse code modulation(PCM), made it possible to move from an analog speech signal to digital.

The numerical value of each sample in this scheme was further represented as a 7-8-bit binary code. The processes of PCM formation are presented in Fig. 8-1, repeated here (see Chapter 1) for clarity and integrity of presentation.

We called this coding codification and made it possible to transmit 256 (2 8) discrete levels of signal amplitude, which was economical for transmission (since it required a central circulation channel with a speed of 64 kbit/s), but not enough from the point of view of voice transmission quality. As a result, before codification, the dynamic range of the voice signal was compressed by a companding circuit (see Chapter 3, Fig. 3-8). This provided speech transmission with a dynamic range of about 42 or 48 dB.

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. Range sound signal occupies a frequency band of 20...20000 Hz.

For enough High Quality(first class broadcast channels) frequency band ∆F C should be 50...10000 Hz, for flawless reproduction of broadcast programs (channels upper class) – 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 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 – Form TV signal on 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 is a pulse close to rectangular in shape. 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. This image will contain maximum amount 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 nominal value line decomposition 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 audio signal 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 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 there are per one horizontal stripe, please note that scanning is 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 to form 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 an 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 power messages 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

The maximum telephone signal power, the probability of exceeding which is negligibly small, is 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 one?

5. How does the spectrum differ? periodic signal from the spectrum of a non-periodic signal?

6. Define signal bandwidth.

7. Explain the essence of fax transmission of messages.

8. How is a TV image scanned?

9. What is the frame rate in a TV system?

10. Explain the principle of operation of the TV transmitting tube.

11. Explain the composition of a complete TV signal.

12. Give the concept dynamic range?

13. List the main telecommunication signals. What frequency ranges do their spectra occupy?

Basic Bandwidth Parameters

The main parameters that characterize the frequency bandwidth are the bandwidth and the unevenness of the frequency response within the band.

The width of the line

Bandwidth is usually defined as the difference between the upper and lower boundary frequencies of the section of the frequency response at which the oscillation amplitude (or for power) is from the maximum. This level approximately corresponds to -3 dB.

Bandwidth is expressed in units of frequency (for example, Hz).

Increasing the bandwidth allows more information to be transmitted.

Frequency response unevenness

The unevenness of the frequency response characterizes the degree of deviation from a straight line parallel to the frequency axis.

The unevenness of the frequency response is expressed in decibels.

Reducing the frequency response unevenness in the band improves the reproduction of the transmitted signal shape.

Specific examples

In antenna theory, bandwidth is the range of frequencies at which the antenna operates effectively, usually the vicinity of the central (resonant) frequency. Depends on the type of antenna and its geometry. In practice, bandwidth is usually determined by the SWR (standing wave ratio) level. SWR METER

In optics, the bandwidth is the reciprocal of the pulse broadening as it passes through optical fiber distances of 1 km.

Since even the best monochromatic laser still emits a range of wavelengths, dispersion broadens the pulses as they propagate through the fiber and thus distorts the signals. When assessing this, the term bandwidth is used. The bandwidth is measured (in this case) in MHz/km.

From the definition of bandwidth it is clear that dispersion imposes a limitation on the transmission range and on the upper frequency of the transmitted signals.