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What is telecommunication system?

By ivykriste   |   16 April 2020

What is telecommunication system?


Communication, technologies, and practices for disseminating information electronically. Centers for modern communication for problems that go far beyond free mail and do not cause damage due to noise and interference. An advanced telecommunication jobs from home must be able to send voice, data, radio and television signals. Digital media are used to achieve reliability and because the cost of digital conversion systems is lower than the cost of analog operating systems. However, to be able to use digital media, analog signals for several audio, radio and telecommunication communications have to be equipped with analog and digital transmissions. (This data is often misplaced because the signs are already in digital form. Most television, radio, and communication systems also use analog systems and are digitized.)


In most cases. The situation, the digitized signal passed through the sources. An adapter that uses a number of techniques to reduce the information. The references are restored and the error code is set to the channel memory code, which contains a large amount of data that can be used to identify and correct errors. The alarm bands are based on transmission using transmitter waves and can be part of a larger signal in a system called multiple messages. The multiplexed beacon is sent to a multi-channel display. Then this extended system turns on the receiving end and then retrieves the information.


Analog-To-Digital Conversion


If the audio, audio or video content is switched on, the material is very reliable - that is, the most reproducible of the original message without the hassle caused by the exchange of sound and noise...

The basis for relatively noise-free and distortion-free telecommunications is a binary signal. Any binary signal that can be used to send messages consists of only two possible values. These values ​​are represented by binary digits or bits, 1 and 0. Unless the noise and distortion accumulated during transmission are significant enough to change the binary signal from one value to another, the receiver can determine the correct cost. So perfect reception can occur.



Analog-to-digital converter begins by sampling or measuring the size of the analog waveform of the individual low-end circuits. The fact that samples of a continuous variable wave can be used to show that the stream is based on the assumption that the flow is limited in its rate of change. Since the communication signal is actually a composite wave - essentially the sum of many components of the sine waves, each of which has its own specific amplitudes and phases - the rate of change of a complex wave can be measured by frequency oscillation of all its components. The difference between the maximum oscillation speed (or the highest frequency) and the minimum oscillation speed (or the lowest frequency) of the sine waves that form a signal is known as the signal bandwidth (B). Bandwidth, therefore, represents the maximum frequency range occupied by the signal. The bandwidth for a sound signal with a minimum frequency of 300 Hz and a maximum frequency of 3,300 Hz is 3000 Gerts or 3 kilohertz. Audio signals typically occupy about 20 kilohertz of bandwidth, while standard video signals occupy approximately 6 million Hertz or 6 megahertz.

The bandwidth concept needed for all communications. Likbavizatsyi analog is a critical assertion that the analog signal does not exceed the individual bandwidth patterns (1 / 2B). This conduit is usually referred to as sampling theory, and the sampling period (1/2 second) is called the neck ist period (named in honor of American electrical engineer Harry Naykista). As an example, in the practice of pre-sampled telephone samples that are symptomatically installed at 3000 Hz, the Nyquist period, at least 1/6000 sec. In today's practice, 8000 samples per second are rummage-sale to increase the frequency of speech and the level of honesty.


To the sample was stored or transmitted digitally, each sample taken from the sample must be reformed into one of an infinite number of values ​​or standards. To facilitate the conversion to binary form, the number of levels is generally of 2 power. Two hundred fifty-six levels are typically used in digital voice transmission as tests have shown that this provides adequate fidelity to the average listener of the phone.


The input to a quantizer is a series of sampled amplitudes for which there is an infinite number of possible values. On the other hand, the output of the quantizer must be restricted to a limited number of levels. Assigning infinitely variable amplitudes to a limited number of levels inevitably introduces an inaccuracy and translates the mistake into the appropriate amount of signal distortion. (For this reason, quantization is often referred to as a "losing" system.) The degree of inaccuracy depends on the number of output levels used by the quantizer. Multiple levels of quantization increase the accuracy of the representation, but they also increase the storage capacity or required transmission speed. Improved performance with the same number of outputs can be achieved by sensitively establishing the output levels and amplitude thresholds required to assign those levels. This arrangement, in turn, depends on the nature of the quantified form. In general, the best quantifier offers more levels at amplitude intervals when the sign is likely to occur and a lower level at which the warning is most suitable. This technique is called a non-linear quantification. Non-linear quantification can also be achieved by passing the signal through the compressor circuit, which increases the weak elements of the message and exceeds its powerful features. The compressed mark, which is now in more dynamic range, can be quantified to homogeneous, linear, threshold and output levels. In the case of a telephone signal, the compression signal is uniformly quantified at 256 levels, and an 8-bit series is shown at each level. At the end of the reception, the reconstructed mark expands to its original amplitude range. This compression and developed sequence, known as the dynamic range combination, can be effective giving 13 bits.


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