NGN Pulse Code Modulation - NGN

What is NGN Pulse Code Modulation?

The emergence of high-speed voice and data communications has created the need for a fast medium for transporting the information. Digital circuits or links have been originated from the need to transmit voice or data in digital form.

Here the conversion from analogue to digital form required a four-stage processes (see the following Figure) explained detailed in the following sections.



Voice frequencies are formed in the variation of an analogue signal i.e. sine wave (see the following Figure). For this , signal has to be converted into a binary form for it to be carried over a digital medium. The basic step of this conversion is to transmit the audio signal into a Pulse Amplitude Modulation(PAM) signal. This is known as sampling.


To perform the sampling process one must collect the required information from the incoming voice frequencies to make a copy of the original signal to be made. Usually voice frequencies are normally between the range of 300Hz to 3400Hz, which are known as the commercial speech band.

Usually to get a sample, a sampling frequency is applied to the original voice frequency. While the sampling frequency is based on the Nyquist Sampling Theorem, which dictates that “the frequency of sampling should be at least twice the highest frequency component.”

You can make ensure that a sample is determined once in each half cycle, by eliminating the possibility of sampling at zero points of the cycle, which would have no amplitude. It will result into the sampling frequency being a minimum of 6.8 KHz.

Here the European standard samples an incoming signal at 8 KHZ, ensuring a sample, is taken every 125micro seconds or 1/8000th of a second (see the following Figure).



Usually the amplitude of any sample can be assigned a binary code (1’s or 0’s), but as there can be an infinite number of amplitudes. So there will be no need to be an infinite number of binary codes available. This process is usually impractical for this another process has to be employed, which is known as quantizing.

Let’s compare the PAM signal against a quantizing scale to Quantizing, which has a finite number of discrete levels. The quantizing scale categorized into 256 quantizing levels, of which, 128 are positive levels and 128 are negative levels.

The quantization stage includes allocating a unique 8 bit binary code appropriate to the quantizing interval into which the amplitude of the PAM signal falls (see the following Figure).


This comprises of 1 polarity bit with the remaining 7 bits used to identify the quantization level (as shown in the above figure).

The first bit is known as the polarity bit, which includes the next three bits for the segment code, which gives eight segment codes, and the remaining four bits for the quantization level, giving sixteen quantization levels.


The quantizing process is known as quantization distortion. This process performed with the sampled signal amplitude falls between the quantization levels. The signal is always rounded up to the nearest whole level. This difference between the sampled level and the quantizing level is quantizing distortion.

Here the percentage of change of the amplitude of a signal differs at various parts of the cycle. It happens most at high frequencies as the amplitude of the signal changes faster than at the low frequencies. To ignore this the first segment code has the quantization levels close together. The next segment code is then double the height of the previous and so on. Here this process called as companding, as it compresses larger signals and expands smaller signals.


In Europe they use the A-law of companding, compared to North America and Japan who use the μ law.

When the quantization distortion is similar to noise then companding improves the signal to noise ratio on low amplitude signals, and produces an acceptable signal to noise ratio over the complete range of amplitudes.


To transmit a digital path for the binary information, you need to modify the information into a suitable line code. The encoding technique employed in Europe is known as High Density Bipolar 3 (HDB3).

HDB3 is originated from a line code called AMI or Alternate Mark Inversion. There are 3 values used in AMI encoding,: no signal to represent a binary 0, and a positive or negative signal that is used alternately to represent a binary 1.


When a long string of zeros are transmitted then a problem associated with AMI encoding occurs which causes phase lock loop problems at the distant end receiver.

AMI and HDB3 works in a same way but it incorporates an extra encoding step that replaces any string of four zeros by three zeros followed by a 'violation bit.’ This violation is of the same polarity of the previous transition (see the following Figure).

You can observe in the following example that 000V replaces the first string of four zeros. When you are using this type of encoding could lead to a mean D.C. level being introduced into the signal, as a long string of zeros could be present, all being encoded in the same way. To overcome this , the encoding of each successive four zeros is changed to B00V, by using a 'Bipolar violation' bit that alternates in polarity.

You can assume that with HDB3 encoding, the maximum number of zeros without a transition is three. This encoding technique is often referred as the modulation format.

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