Telecommunication in the 21st Century
Telecommunication in the 21st century have improved over the decade by the introduction of better techniques through which signals can be transmitted from a transmitter through a medium to a receiver. These techniques have improved mobile communications, satellite transmissions and helped to improve data security. Some of these techniques are amplitude modulation (AM), frequency modulation (FM), sampling and link analysis (SLA) and PCM.
The acronym PCM represents ‘Pulse-code modulation’, which is used for digitizing analogue data, for instance, audio signals. This is carried out by sampling analogue signals at uniform interval and then quantized to a series of symbols in a digital code (e.g. 10001).Its technically a way in which analogue signals are converted to digital form. PCM technique has its advantages;
- It makes processing of signals cheap since PCM is digital.
- It helps to filter off frequencies above the highest signal frequency.
Get your grade
or your money back
using our Essay Writing Service!
Pulse-code modulation has been a form used for some compact disc formats, digital video and for digital audio in computers.
In PCM, there are series of processed to be followed;
- Binary coding
This is a process where frequencies above the highest signal frequency are removed. The reason for this is that if this frequency is not removed, problems would occur when going to the next stage of sampling.
This stage of the PCM is performed through PAM (pulse amplitude modulation).It answer the question of how signals change from one form to another (analogue to digital).
It makes use of the original analog signal and uses it for the amplitude modulation of a pulse which has constant amplitude and frequency, this constant frequency is known as the sampling frequency (i.e. the number of samples per second ).The sampling frequency have to be more than the maximum frequency of the analogue signal. To work out the sampling rate, Nyquist theorem is used;
“That in order to be able to reconstruct the original analogue signal, a minimum number of samples had to be taken”.It could be stated as:
Fs > 2(BW)
Fs = Sampling frequency
BW = Bandwidth of original analog voice signal
Quantizing and Coding
This basically means the converting of each of the analogue sample into a discrete value (in the form of a binary code) that can be given a digital code word. It is done by assigning each sample a certain quantization interval. The instantaneous amplitude is been rounded off to certain levels, this thereby introduces some uncertainties (quantization noise).This is given by this expression;
Number of levels = 2 ^ Bn (Bn is the number of bits used in the encoding)
It was proven from the experiment that the higher the number of quantization levels the lesser the amount of quantizing noise. However this process of increasing the quantizing level to lower the quantizing noise introduces complexity into the system as the PCM system would need to be able to handle more code word.
It is a word derived from the combination of compressing and expanding. This is another stage in pulse–code modulation. It is a process of compressing a given analogue signal and this signal is expanded to its original size on getting to destination. In this process, the input signal is compressed into logarithmic segments and then quantized and coded. The more the signals increase the more the compression increases.
Since the larger signals are compressed more than the smaller signals, the quantization noise increases. This indirectly keeps the SNR (signal to noise ratio) constant.
EXPRERIMENTATION AND OBSERVATION
- PCM ENCODER module
- Connection cable
The experiment was carried out by sending an input (analogue message) into the PCM ENCODER module. This input is constrained to a defined bandwidth and amplitude range in order to make sure the Nyquist criterion is observed. The PCM ENCODER module looks like the diagram below:
Always on Time
Marked to Standard
A suitable encoding scheme for the analogue sample is selected. For example a 4-bit or 7-bit encoding scheme. The analogue signal is fed through the Vin. For this experiment, the clock rate us 8.33 kHz TTL signal from MASTER SIGNAL module.
Time frame is also very essential as each binary word is located in a time frame. It’s 8 clock periods long and has 8 slots of equal length (i.e. 0 – 7). The LSB (consisting of 1’s and 0’s) are embedded in the encoder itself. This is useful in determining the location of each frame in the data stream. Initially the 4-bit linear coding scheme is selected and patched up with the 8.33 kHz TTL sample clock.CH-2A displays the clock signal on the oscilloscope. The display below shows a 4-bit PCM output for zero amplitude input;
Quantization in PCM ENCODING is the next stage after sampling. The quantization level is rather transmitted instead of the sample value. The quantization levels are binary coded (i.e. binary ‘1’ in the presence of a pulse and binary ‘0’ in the absence of a pulse)
RESULTS AND OBSERVATION
The output of the variable DC is connected to Vin and sweeping the DC voltage slowly forward and backward shows discrete jumps in the data pattern, e.g.
The maximum voltage is recorded as -2.51V.Also increasing the amplitude of the DC input signal looks like the diagram below;
Changing the DC voltage from the maximum to minimum gave a range of binary code variations as listed below;
The following measurements were later made after recording the quantizing levels and associated binary numbers;
- Sampling rate – 16.6 kHz
- Frame width – 950µs
- Width of a data bit - 120µs
- Width of a data word - 480µs
- Number of quantizing level – 16
From the measurement above it could be concluded that the quantizing levels are linearly spaced .The same process would be applicable to a 7-bit linear encoding using the toggle switch on the front panel, though it would take longer than the 4-bit linear encoding done earlier.
The Companding stage in a PCM is the process by which an analogue signal is been compressed at the source and then expanded back to its original size when it gets to its destination. During this process, the signal is compressed into segments which are quantized using uniform quantization. As the sample signal increases, the compression increases (i.e. the larger samples gets more compressed than the smaller samples). The standard of companding used in this experiment is the A-law .The equation is;
Where A = 87.7 in Europe and X is the normalized integer to be compressed.
RESULTS AND OBSERVATION
The toggle switch is changed back to a 4-bit companding and the TIMs A4 companding law pre-selected is selected from the switch board. This gave the measurement below;
In PCM decoding, the TIMs PCM DECODER module is used for decoding. This is the first operation in the receiver towards regenerating the received pulses. Amplitude of the pulse generated is the linear sum of all pulses in the coded word. In other to be able to recover the information on the PCM decoder, the knowledge of the sampling rate used to encode the signal is essential.
RESULTS AND OBSERVATION
The setup is similar to the earlier setup with CH-1A connected to the scope selector to the PCM output of the PCM ENCODER.A large negative DC is used for the message, the alternating ‘0’ and ‘1’ bits produced are measured to be 1920ms apart. The 4-bit linear decoding scheme is now selected to carry out the decoding process. The 8.33 kHz TTL signal is stolen from the transmitter and connected to the clock input.
Time division multiplexing (TDM) is an alternative to the method of multiplexing using frequency sharing. Each channel is allocated a specific time slots, and each slots contain frames which must be repeated at the sampling rate. It can only be used for pulsed signals and not for analogue signals because they are continuous in time. The importance of TDM is that it enables many independent signals to be transmitted.
This Essay is
a Student's Work
This essay has been submitted by a student. This is not an example of the work written by our professional essay writers.Examples of our work
RESULTS AND OBSERVATION
A PCM TDM signal could be generated using PCM ENCODER; each driven by the same clock ( one the MASTER and the other SLAVE).Interconnecting in this way eliminates other frames and gives room for the two output to be added together to form the TDM signal. The display on the oscilloscope is shown below;
The connection of the MASTER and the SLAVE generates the diagram below;
Patching up the two PCM data outputs generates the display below;
The next step which is shown below is to confirm that the frame synchronization bit is a ‘1’ for the MASTER and ‘0’ for the SLAVE
The last stage of this experiment is to separate the two messages that have been multiplexed earlier. The PCM demodulator is patched up, with each module receiving the same clock stolen from the transmitter and each module also receives an external FS signal. The diagram below confirms the two messages have been recovered and appear at the correct outputs;
Pulse Code Modulation is however a very effective way of conveying audio signal by sampling the signal and transmitting binary coded pulse representing the sample values. It has emerged the most favored modulating scheme for transmitting analogue information such as voice and video signals. The advantages of PCM over the other forms of modulation (e.g. analogue modulation) are;
- PCM suppresses wideband noise.
- It is effective in the regeneration of the coded signal along the transmission path.
- It enables digital multiplexing.
- It enables the efficient exchange of increased channel bandwidth for improved signal-to-noise ratio.
All these advantages however come at the expense of increased system complexity and increases channel bandwidth.
http://www.webopedia.com/TERM/P/PCM.html [last accessed 25/03/08]
[last accessed 25/03/08]
[last accessed 25/03/08]
http://www.comlab.hut.fi/opetus/245/2004/09_PCM.ppt#20 [last accessed 25/03/08]
Rodger E.Ziener and William H.Tranter, “Principles of Communication”, Chapter 3, John Wiley and sons, NY, 2002.
Simon Haykin, “Communication Systems”, Chapter 3, John Wiley and sons, NY, 2001.
David Petersen, “Audio, Video and Data Telecommunications”, Chapter 2, McGraw-Hill, Cambridge, 1992.