Method of Speech Signal Compression in Speaker Identification Systems

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In this paper, the authors present a technique of efficacy improvement of speech signal compression algorithm without individual features speech production loss. The compression in this case means to delete, from the digital signal, those quantization steps that can be predicted. The authors propose to decrease the number of those quantization steps using a modified linear predication algorithm with variable order. That allows to decrease compression time and save computer resource. The task of efficient representation of speech signal is one of the vital tasks in speaker identification problems. For example, an automatic speaker recognition system is installed on a LAN or WAN server, which authorizes a terminal to access the network according to the voice of the subscriber.