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Pulse-code modulation

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Pulse-code modulation
NamePulse-code modulation
CaptionSimplified diagram of PCM encoding
Invented1937
InventorAlec Reeves

Pulse-code modulation. Pulse-code modulation (PCM) is a foundational method for digitally representing sampled analog signals, forming the bedrock of modern digital audio and telecommunications. It operates by regularly measuring the amplitude of an analog signal and quantizing each sample to the nearest value within a fixed range. This process yields a stream of binary numbers that can be efficiently stored, transmitted, and reconstructed. The fidelity of the reproduced signal is primarily governed by the sampling rate and the bit depth used during the encoding process.

Overview

PCM serves as the standard form for digital audio in applications ranging from compact discs to computer sound files and digital telephony systems like the T-carrier. Its development was a pivotal step in the transition from analog to digital communications, enabling robust, noise-resistant signal transmission. The technique's core principle involves two distinct stages: sampling and quantization, which convert a continuous waveform into a discrete digital representation. This digital format is essential for processing by modern devices such as digital signal processors and is a prerequisite for many forms of data compression.

Technical description

The PCM process begins with a sample and hold circuit that captures instantaneous voltage levels of the input signal at uniform intervals defined by the Nyquist–Shannon sampling theorem. Each captured sample's amplitude is then mapped, or quantized, to the nearest level in a predefined set, a step that introduces a small, inherent error known as quantization noise. This quantized value is subsequently encoded into a binary code, typically using a method like two's complement for signed values. Key parameters include the sampling frequency, which must exceed twice the highest audio frequency present to avoid aliasing, and the bit depth, which determines the dynamic range and signal-to-noise ratio of the encoded audio.

History and development

The concept of PCM was invented and patented in 1937 by British engineer Alec Reeves while working at the International Telephone and Telegraph corporation. Practical implementation was delayed for decades due to the complexity of required circuitry, with significant early work conducted at Bell Labs in the post-war era. A major breakthrough came with the invention of the transistor, which made the dense, high-speed switching circuits for PCM economically feasible. The first widespread application emerged in the Bell System's T1 carrier system in 1962, which digitized voice channels for multiplexed transmission. Later, PCM became the standard encoding for the compact disc, developed jointly by Philips and Sony, cementing its role in consumer audio.

Applications

PCM is the direct encoding format for audio on compact disc digital audio, DVD-Video, and much computer-based LPCM audio, such as that in WAV and AIFF files. It forms the core of digital telephony networks, including the E-carrier and Synchronous Optical Networking hierarchies. In professional audio, interfaces like AES3 and MADI transport multichannel PCM data between studio equipment. Furthermore, it acts as the uncompressed source signal for various lossy compression codecs, including those standardized by the Moving Picture Experts Group and used in broadcasting systems like Digital Audio Broadcasting.

Limitations and variants

Standard PCM's primary limitations include its high bit rate, which demands significant storage and bandwidth, and the inherent trade-off between dynamic range and quantization noise. To address these, numerous variants and related techniques have been developed. Differential pulse-code modulation reduces bit rate by encoding the difference between consecutive samples. Adaptive differential pulse-code modulation, used in early digital telephony like the G.726 standard, varies the quantization step size based on the signal. In the audio domain, pulse-density modulation is the underlying principle for Direct Stream Digital encoding used in Super Audio CD. Other related modulation schemes include delta modulation and its adaptive form, which use one-bit code words to track signal changes.

Category:Digital signal processing Category:Telecommunication theory Category:Audio codecs