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PCM in Satellite Radio

The Pulse Code Modulation was excelled by the British engineer Alec Reeves. Very first transmission of a message using PCM was the SIGSALY voice encryption equipment used in high-level Allied communications during World War II. This original primary application for PCM was to convert analog signals into digital format by taking samples of wave forms that fall within 8 to 192 KHz and converting them into a digital number 8 to 24 bits long. In a nutshell, PCM refers to the technique of digitalizing an analog signal by sampling the magnitude of the signal at uniform intervals and converting it into a series of digital or binary code. Next to the receiving end, a pulse code demodulator converts these binary numbers into pulses with the same quantum levels as in the modulator, which are then further processed to obtain the original analog waveform.

The PCM was originally intended for use in telephone systems, but in the 21st century, it is also the standard way for digitalizing analog data such as in digital audio, digital video and CD formats, telemetry and virtual realty. Inside conventional pulse code modulation, the analog signal can be processed before it is digitalized and it is not, however, subjected to further processing before being multiplexed into the aggregate data stream. Few types of pulse code modulation combines signaling as well as coding. Basic difference is that while its original application was in the analog domain, newer applications find it useful in the digital domain. Though, the transformed based signal compression techniques that are followed today have rendered these obsolete, and are far more efficient. The differential pulse code modulation or delta pulse code modulation is a technique in which PCM values are encoded as the difference between the current and previous values.

Within audio formats, this type of encoding reduces the number of bits required per sample by about 25% compared with PCM. An additional variant of DPCM is known as ADPCM or Adaptive PCM, which varies the type of quantization step, thereby allowing further reduction of bandwidth for a given signal-to-noise ratio. And in telephone communications, the standard audio signal is encoded in a digital format known as DSO. In North America and Japan, the default encoding on a DSO is law PCM. And in Europe and most of the rest of the world, it is A-law PCM. This logarithmic compression system is described by international standard G.711. Rider the circuit costs are high and you don't mind a slight loss in voice signal quality then further audio compression is possible using the ADPCM algorithm. The technique is detailed in G.726 standard; however, some existing technologies are covered by privately owned patents, which mean that the end user may have to pay the patent holder in order to use their technology.

About the Author

Tymon Hytem has worked in the electronics field for the past 15 years. He enjoys helping people decide on electronic gadgets from telephones to XM Radio and choosing the perfect XM Satellite Radio system for their needs.

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