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PCM (Pulse Code Modulation)

Updated: Oct 17, 2023

Pulse Code Modulation(PCM) Block Diagram, Working Principle

Pulse Code Modulation(PCM) is a digital modulation technique where the analog signal is coded into digital pulses. Here, we will see the block diagram of the PCM system or Pulse Code Modulation system to understand its working principle. For a long transmission system, a carrier signal is used and according to the actual data-carrying signal, the carrier signal is modulated.

So, in simple words, the process of varying the parameters of a carrier signal is known as modulation. There are various types of modulation techniques are available related to analog and digital systems. The PCM or Pulse Code Modulation is one of them.

Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the analog-to-digital process; newer implementations do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques, such as modified discrete cosine transform (MDCT) coding.

  • Linear PCM (LPCM) is PCM with linear quantization.

  • Differential PCM (DPCM) encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM.

  • Adaptive differential pulse-code modulation (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio.

  • Delta modulation is a form of DPCM that uses one bit per sample to indicate whether the signal is increasing or decreasing compared to the previous sample.


If we know the function of each part of the system then we can easily understand the working principle of PCM. Now let's see the important parts or blocks of the PCM system are,


As its name suggests, the low-pass filter will pass only low-frequency signals. So, the analog signal is applied to the input of the low pass filter and the low pass filter will block the high-frequency signal and pass the low-frequency signal to the next stage. A frequency is set as per the frequency of the message signal in the filter circuit with respect to which the signal will be cut off.


The sampler circuit helps to take sample data from the message signal at the instantaneous value. It can be called a process of cutting the continuous signal into discrete from. The low-frequency analog signal comes from the filter circuit to the sampler circuit and it is sampled at regular intervals.

Sampling is based on the Sampling Theorem, which is based on the fixed sampling rate, called the Nyquist rate. Hence, the sampling theorem is also known as the Nyquist theorem. It is based on the theory of bandlimited signals.

Analog signals have a frequency, or how many times they go up and down in a second. This frequency is measured in hertz. Claude Shannon's explanation of the theorem is: "If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart." To accurately reproduce a signal, the sample rate must be twice the highest frequency.

Generally, according to Nyquist–Shannon sampling theorem the sampling rate or sample frequency is twice the highest frequency component of the message signal. So, fs ≥ 2fm

Here, Fs= sampled frequency, and Fm = message signal frequency.


The quantizer is a circuit that does quantizing or compresses or reduces the no of samples. It removes the excessive bits from the sampled signal provided by the sampler circuit. Basically, it confines the data and makes rounds of each sample. Actually, it gives the shape of digital bis or digital pulse from the sinusoidal shape. For example, if the voltage is 5V then it is considered as high but if the voltage is 0V, then it is considered as low. So the quantizer can make the 4.8V to 5V or 0.5V to 0V or remove the 3V signals.

There are two types of quantizing processes - uniform quantizing and non-uniform quantizing. In the uniform quantizing process there is a uniform spacing between the levels whereas, in the non-uniform quantizing, the spacing between the levels is non-uniformed.


The encoder circuit converts the quantizing signal into binary codes. So this unit generates the actual digitally encoded signals. This digital signal consists of binary pulses and they act as modulated output from the transmitter section of the whole system.


The regenerative repeater is placed at both ends of the transmission channel sending and receiving end. at the sending end, it makes ready the signal for long-distance transmission. And at the receiving end, it removes noise from the signal and gives reshape the signal to get the actual signal. There may be multiple numbers of regenerative repeaters for long-distance transmission.


A transmission channel is a medium or path for data transmission. Through this channel or medium data or signal is transmitted.


At the receiving end of the transmission channel, the regenerator circuit is used. It also works the same as the regenerator repeater. It removes the noise and distortion from the signal received from the transmission channel and generates the actual signal sent.


The decoder circuit converts the received digitally coded binary signal into the original signal. This circuit also demodulates the signal. The generative circuit and decoder both are responsible to convert the received digital signal into an analog signal.


The filter at the receiving end of the system is known as the reconstruction filter. The reconstruction filter provides the actual original signal sent by the sender.

Digital signals take the form of two states that can be represented by 0s and 1s, but this information needs to be arranged in a specific way to be of any use. In just about every case, the system to organize the digits is known as PCM (Pulse Code Modulation).

With PCM, the original analog music waveform is described in two parts.

The first is its amplitude (size). In CD this is represented by 16 bits of digital data, which gives us the ability to define 65,536 different signal levels.

The original music waveform has to be measured at regular intervals in order for it to be represented properly. The measurement is done 44,100 times a second. While that looks like an arbitrarily large number, it’s chosen quite carefully to ensure that the full frequency range of human hearing (20Hz to 20kHz) is covered.

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