Abstract
The field of speech compression has advanced rapidly due to cost-effective digital technology and diverse commercial applications. In voice communication a real-time system should be considered. It is not still possible to compress signals without facing any loss in real-time system. This paper presents a theory of loss-less digital compression for saving high quality speech signals. Emphasis is given on the quality of speech signal. In hearing music high quality music is always needed, consuming smaller memory space. In this compression 8-bit PCM/PCM speech signal is compressed. When values of samples are varying they are kept same. When they are not varying the number of samples containing same value is saved. After compression the signal is also an 8-bit PCM/PCM but expansion is needed before hearing it. This technique may also be used in real-time systems.
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