Contents I. Introduction
This paper discusses VLSI implementation of speech digitization technique used by many telecommunication companies. The main reason for digitizing an analog signal is for the computer to understand. The cost of processing, transmission and storage costs are higher and also with low portability for analog signals. The analog signals are digitized with respect to amplitude and time. The conversion of time is known as sampling, which is a process of generating discrete samples by converting the continues process of the analog signal. The lowest frequency which we can sample the data is limited by the Nyquist criterion. This is the first step of digitizing the signal, secondly the process of converting the amplitude is known as quantization. In quantization the discrete amplitude signal is approximated to its next closest signal, the resulting signal is then divided into partitions. Non uniform quantization is used in order to reduce the number of bits. Once, this is completed the final step is the bit assignment, the signals are represented binary digit form. This is known as PCM (Pulse code Modulation). PCM does not consider the previous samples therefore it is a memory free technique. PCM works on companding techniques such as A-Law and U-Law.
