Pulse code modulation


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Pulse code modulation

  1. 1. Pulse Code Modulation Pulse Code Modulation (PCM) Is a Digital Scheme for Transmitting Analog Data. The Signals in PCM Are Binary; That Is, There Are Only Two Possible States, Represented by Logic 1 (High) and Logic 0 (Low). This Is True No Matter How Complex the Analog Waveform Happens to Be. Using PCM, It Is Possible to Digitize All Forms of Analog Data, Including Full-motion Video, Voices, Music, Telemetry, and Virtual Reality (VR).
  2. 2. Why PCM The stream of pulses and non-pulse streams of 1’s and 0’s are not easily affected by interference and noise. Even in the presence of noise, the presence or absence of a pulse can be easily determined. Since PCM is digital, a more general reason would be that digital signals are easy to process by cheap standard techniques. This makes it easier to implement complicated communication systems such as telephone networks.
  3. 3. Introduction <ul><li>PCM was invented by P. M. Rainey of Western Electric in 1926 and later improved by British engineer Alec Reeves in 1937 while working for International Telephone and Telegraph in France. He filed for a French patent in 1938, and his U.S. patent was granted in 1943. </li></ul>
  4. 4. Nomenclature <ul><li>The word pulse in the term pulse-code modulation is somewhat confusing, as there appear to be no &quot;pulses&quot; per se anywhere to be found except in the transmission line. This perhaps is a natural consequence of this technique having evolved alongside two analog methods, pulse width modulation and pulse position modulation, in which the information to be encoded is in fact represented by discrete signal pulses of varying width or position, respectively. In this respect, PCM bears no resemblance to these other forms of signal encoding, except that the binary numbers of the PCM codes are represented as electrical pulses. The device that performs the coding and decoding function in a telephone circuit is called a codec. </li></ul>
  5. 5. Implementation <ul><li>The practical implementation of PCM makes use of other processes. The processes are carried out in the order in which they appear below: </li></ul><ul><li>Sampling </li></ul><ul><li>Quantizing </li></ul><ul><li>Encoding </li></ul>
  6. 6. Topics of Discussion <ul><li>To obtain PCM from an analog waveform at the source (transmitter end) of a communications circuit, the analog signal amplitude is sampled (measured) at regular time intervals. The sampling rate, or number of samples per second, is several times the maximum frequency of the analog waveform in cycles per second or hertz. </li></ul>
  7. 7. Topic One-Sampling <ul><li>Details about this topic </li></ul><ul><li>Supporting information and examples </li></ul><ul><li>How it relates to your audience </li></ul>
  8. 8. Topic Two-Quantization The instantaneous amplitude of the analog signal at each sampling is rounded off to the nearest of several specific, predetermined levels. This process is called Quantization.
  9. 9. Topic Three-Encoding <ul><li>The number of levels is always a power of 2 -- for example, 8, 16, 32, or 64. These numbers can be represented by three, four, five, or six binary digits (bits) respectively. The output of a pulse code modulator is thus a series of binary numbers, each represented by some power of 2 bits. </li></ul>
  10. 10. Application <ul><li>Supporting information and examples- Digital information is better transmitted in its digital form because converting the signal to analog and sending it through an analog network can be costly. Digital data is easily compressed; therefore it can be transmitted using a small bandwidth. Because of the nature of devices used to boost the signal strength during transmission, error performance is much improved when compared with analog. It is also better to transmit information in digital form because computer components used in the transmission process are very reliable. </li></ul>
  11. 11. Real Life <ul><li>The microphone and line-in circuits on a sound card generate PCM samples, and all sound cards require PCM for output. Compressed audio formats such as MP3 and AAC are converted to PCM first, and The sound card converts the PCM to analog for the speakers. </li></ul>
  12. 12. LIMITATIONS <ul><li>It may be noted that there are two sources of impairment implicit in any PCM system: </li></ul><ul><li>choosing a discrete value near the analog signal for each sample (Quantization error) </li></ul><ul><li>between samples no measurement of the signal is made; due to the sampling theorem this results in any frequency above ( f s being the sampling frequency) being distorted or lost completely. </li></ul>
  13. 13. END Thank you