Pulse Code Modulation (PCM) Working Principle

Working Principle

Pulse Code Modulation converts a time-continuous, value-continuous analog signal into a time-discrete, value-discrete digital signal for transmission over a channel. Pulse Code Modulation is the process of first sampling the analog signal, then quantizing and encoding the sampled amplitude values.
Sampling is the periodic scanning of an analog signal, converting a time-continuous signal into a time-discrete signal. After sampling, the analog signal must still contain all the information of the original signal, meaning the original analog signal can be recovered without distortion. The lower limit of the sampling rate is determined by the sampling theorem, and the sampling rate is 8 kbit/s.
Quantization is the process of discretizing the amplitude of the instantaneous values obtained from sampling, i.e., using a set of specified levels to represent each instantaneous sampled value with the nearest level.
After an analog signal is sampled and quantized, a quantized pulse amplitude modulation signal is obtained, which has only a finite number of values.
Encoding is the process of using a set of binary code groups to represent each quantized value with a fixed level. In practice, quantization is completed simultaneously during the encoding process, so the encoding process is also referred to as analog-to-digital conversion, denoted as A/D.

Pulse Code Modulation, abbreviated as PCM, first passes the voice signal through an anti-aliasing low-pass filter, then performs pulse sampling to produce a sampling signal with an 8 kHz repetition rate (i.e., a discrete pulse amplitude modulated PAM signal). The amplitude-continuous PAM signal is then quantized into a signal with a finite number of amplitude values using a "rounding" method, and subsequently encoded into binary codes. For telephony, CCITT specifies a sampling rate of 8 kHz, with each sampled value encoded into 8 bits, i.e., a total of 2^8 = 256 quantized levels. Therefore, the standard digital rate after PCM encoding for each voice channel is 64 kb/s. To address the issues of large quantization errors and poor voice quality for small signals in uniform quantization, a nonlinear quantization method with non-uniform quantization intervals is adopted in practice. That is, the quantization characteristic has denser levels and smaller intervals for small signals, while for large signals, the levels are sparser and the intervals are larger.

In practice, two logarithmic compression characteristics are used: A-law and μ-law. A-law encoding is mainly used in 30/32-channel primary multiplex systems, while μ-law encoding is mainly used in 24-channel primary multiplex systems. A-law PCM is used in Europe and China, while μ-law PCM is used in North America and Japan.

Pulse Code Modulation - Encoding

PCM encoding principle and rules: The PCM digital interface follows the G.703 standard, transmitting asymmetrically or symmetrically over 75 Ω coaxial cable or 120 Ω twisted pair. The transmission line code is HDB3 code, which contains timing information. The receiving end can recover timing through decoding to achieve clock synchronization. Fb is the frame synchronization signal, and C2 is the clock signal with a rate of 2.048 Mbps. Data is valid on the falling edge of the clock. The E1 interface has a PCM frame structure: one multiframe consists of 16 frames, and one frame is 125 μs, divided into 32 timeslots. Among these, timeslot 0 of even frames transmits the synchronization information code 0011011, timeslot 0 of odd frames transmits the alarm indication code, timeslot 16 transmits signaling information, and the other timeslots transmit data. Each timeslot transmits 8 bits of data.