Pulse Code Modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a digital (usually binary) code.
Figure 1. Block diagram of a basic PCM system
Key Concept: PCM is the standard form of digital audio in computers, compact discs, digital telephony, and other digital audio applications.
PCM was invented in 1937 by Alec Reeves at the International Telephone and Telegraph. The technology to implement it practically wasn't available until the development of semiconductor components which started in 1948. The first successful commercial PCM telecommunication system was developed at Bell Labs and put into operation in 1962. This system, known as T1, utilized 24 digital channels and was a significant advancement in telecommunications, enabling greater capacity through digitization and time-division multiplexing.
PCM transmitter involves four main steps:
Low Pass Filter eliminates the high frequency components present in the input analog signal which is greater than the highest frequency of the message signal, to avoid aliasing of the message signal.
Sampling converts a continuous-time signal into a discrete-time signal by measuring the signal's amplitude at regular time intervals.
The sampling theorem states that a signal can be exactly reconstructed if the sampling frequency is greater than twice the highest frequency component in the signal.
Where:
Example: For audio signals with a maximum frequency of 20 kHz, the sampling rate should be at least 40 kHz. CD-quality audio uses 44.1 kHz.
Quantization converts the continuous-amplitude samples into discrete amplitude values.
The number of possible amplitude values is determined by the number of bits used:
Where n = number of bits per sample
| Bits per sample | Quantization levels | Application |
|---|---|---|
| 8-bit | 256 | Telephone quality |
| 16-bit | 65,536 | CD audio |
| 24-bit | 16,777,216 | Professional audio |
The difference between the actual analog value and the quantized digital value is called quantization error or quantization noise.
Each quantized sample is converted to a binary code. The most common encoding method is linear PCM where the quantization levels are uniformly spaced.
For a 3-bit system (8 levels):
| Quantization Level | Binary Code |
|---|---|
| 0 | 000 |
| 1 | 001 |
| 2 | 010 |
| ... | ... |
| 7 | 111 |
The PCM receiver consists of three main parts:
1. Regenerator: A regenerative repeater is placed at the receiving end also so as to have an exact PCM transmitted signal. Here, also the regenerator works in a similar manner as that when employed in the transmission path. It eliminates the channel induced noise and reshapes the pulse.
2. DAC and Sampler: Digital to analog converter performs the conversion of digital signal again into its analog form by making use of the sampler. As the actual message signal was analog thus at the receiver end there is a necessity to again convert it into its original form.
3. LPF: The sampler generates analog signal but that is not the original message signal. Thus, the output of the sampler is fed to the LPF having cutoff frequency fm. This is sometimes termed as the reconstruction filter that produces the original message signal.
Encodes the difference between the current sample and a predicted value from previous samples.
Varies the size of the quantization step to allow better reproduction of signals with varying amplitude.
The bit rate of a PCM system is calculated as:
Example: CD-quality audio has:
Bit rate = 44,100 × 16 × 2 = 1,411,200 bits/sec (1.411 Mbps)
Answer: According to Nyquist theorem, it should be greater than 30 kHz (2 × 15 kHz).
Answer: 212 = 4,096 levels.
Answer: 6.02 × 16 + 1.76 = 98.08 dB.
Answer: 8,000 × 8 × 1 = 64,000 bits/sec (64 kbps).
Answer: Any two from: noise immunity, regeneration capability, flexibility, security, easier storage.