AbstractThis research is concerned with the development of a system designed to economise on the bandwidth required for the digital transmission of speech signals.The sampling rate in the PCM system is constant and related to the maximum frequency of the sampled signal, then, at times when lower frequencies only are being transmitted, the sampling rate is unnecessarily high, and there would thus appear to be the possibility of reducing the bandwidth. This led to the thought of a new method of sampling. The new method has been called pulse time-code modulation (PTCM).The sampling in PTCM is achieved by dividing the full range of the signal amplitude into quantizing levels. When the signal crosses one of these levels the time measurement starts. It continues until another crossing of the adjacent level occurs, then a digital word is completed and a new measurement of time starts for a new digital word. Every digital word consists of one bit indicating the direction of crossing up or down and the rest of the word represents the measurement of time. The transmitting rate should be the average of the sampling rate. This can be done by storing the samples and averaging their rate. At the receiving end the process is reversed.
It was found that it is possible to achieve appreciable saving in transmission rate. A large store is necessary for the purpose of averaging. The presence of such a store will necessarily cause a delay in the transmitted signal, which is a disadvantage when the PTCM system is used for duplex transmission.
During the course of the research into the above system, it became apparent that the same principles of averaging could be applied to the conventional PCM system. This was investigated, and it was found that an appreciable saving in the bandwidth can be achieved. We have called this system AVERAGE RATE PULSE CODE MODULATION (ARPCM)
|Date of Award||31 Dec 1981|
- pulse time-code
- modulation system