Pulse code modulation (PCM) is a digital modulation technique by which analog signal gets converted into digital form for transmission, storage, or processing. It involves sampling, quantizing, encoding, and, if needed, reconstructing the original analog signal.
Basics of Pulse Code Modulation (PCM)
Pulse Code Modulation (PCM) is a digital scheme, that digitize all forms of analog data, including video, audio, music, telemetry, etc. In PCM, the continuous analog signal is discretized into discrete values, which are then encoded into binary numbers.
This technique is widely used in various applications, such as telecommunications, audio recording, and data transmission, to convert and transmit analog information in a digital format, It allow efficient storage, transmission, and processing of signals while maintaining a high level of fidelity.
Pulse code modulation (PCM) is a digital modulation technique where as PPM (pulse position modulation), PWM (pulse width modulation) are the example of analog modulation techniques.
Read more: Block Diagram of Digital Communication
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Block Diagram of PCM
A communication system consist of three section a transmitter, communication channel, and receiver. A transmitter and a receiver have various components depending on the input signal and the output requirements. Transmitter perform modulation and receiver perform demodulation functions.
In modulation process, sends the message signal with the carrier signal, which helps in enhancing the signals’ characteristics. It also removes any noise, interference, or distortion in the signal. The demodulation process recovers the original signal to make it suitable for the receiver.
The block diagram of the Pulse Code Modulation (PCM) system is shown below:
1. Transmitter Section: The transmitter section consist of low pass filter, sampler, quantizer, encoder. The function of all the component explained below.
Analog Signal Input: PCM starts with an analog signal as its input. This analog signal can represent various types of data, such as audio waveforms in the case of voice or music.
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LPF (low pass filter): A Low Pass Filter (LPF) passes all the low frequency and rejects the higher frequencies from the input signal. It is done to avoid the problem of aliasing or distortion in the input signal.
Sampler: Sampling refers to the process of converting continuous time signal into discrete form this is done by sampler.
In this process the continuous time signal (analog signal) is sampled at regular time intervals. Each sample take the instantaneous amplitude of the analog signal at that moment. The rate at which these samples are taken is called the “sampling rate” or “sampling frequency.” According to Nyquist’s theorem, the sampling rate must be at least twice the highest frequency present in the analog signal to avoid aliasing.
The output of sampler is a discrete time signal.
Quantizer: After sampling, each sample’s amplitude is quantized using quantizer. This involves dividing the range of possible amplitudes into a finite number of discrete levels. The number of quantization levels is determined by the “bit depth” or “resolution.” Common bit depths are 8 bits, 16 bits, or 24 bits. More bits allow for finer amplitude resolution.
Encoder: The digitization of analog signal is done by the encoder. Each quantization level is assigned a unique binary code or digital word. The length of this binary code is determined by the bit depth. For example, with 8-bit PCM, there are 256 quantization levels, each represented by an 8-bit binary code.
2.Communication channel: A communication channel is a medium between the transmitter and the receiver. The PCM bitstream can be transmitted over digital communication channels. It also includes a repeater and regenerator that can regenerate the coming digital signal, increase signal strength, and reduce effect of noise.
3. Receiver section: The receiver section consist of decoder, Reconstruction Filter. The function of all the component explained below.
Decoder: Decoder perform the opposite operation of encoder placed in the transmission section. The digitally encoded signal arrives at the receiver. It first removes the noise from the signal. Then decoder circuit decodes the receive pulse coded waveform to reproduce the original signal. This circuit acts as the demodulator.
Reconstruction Filter: To smoothen the reconstructed analog signal and remove high-frequency components introduced during quantization, a reconstruction filter is often used.
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What is PCM, and how does it work?
PCM stands for Pulse Code Modulation. It’s a method of digitally encoding analog signals by sampling the signal’s amplitude at regular intervals, quantizing the samples into binary values, and then encoding them into a digital bitstream.
What is the purpose of sampling in PCM?
Sampling in PCM involves measuring the amplitude of an analog signal at specific time intervals. This process converts the continuous analog signal into discrete data points for digital representation.
What is quantization, and why is it necessary in PCM?
Quantization involves assigning a discrete value (usually binary) to each sampled amplitude. It’s necessary to represent the analog signal accurately in digital form. The number of quantization levels determines the bit depth, which affects the signal’s resolution.
What is the Nyquist Theorem, and how does it relate to PCM?
The Nyquist Theorem states that the sampling rate must be at least twice the highest frequency component in the analog signal to accurately reconstruct it from its samples. PCM adheres to this principle to avoid aliasing.
What are the advantages of using PCM for audio recording?
PCM offers high-fidelity audio representation, compatibility with a wide range of devices and software, robustness against noise, and flexibility in terms of adjusting audio quality.