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Analysis and Synthesis of Speech
In modern digital systems, a speech signal is represented in a digital format that is comprised of a sequence of binary bits. For storage and transmission applications, it is often desirable for the speech signal to be represented in as few bits as possible, while maintaining its perceptual quality.
In narrowband digital speech compression, digital speech signals are sampled at a rate of 8000 samples per second. Typically, each sample is represented by 8-bits. This corresponds to a bit rate of 64 kbits per second. Further compression is possible at the cost of quality. Most of the current low bit rate speech coders are based on the principle of linear predictive speech coding. The simplest implementation of this compression technique is presented in the linear prediction coefficient (LPC) Analysis and Synthesis of Speech demo. This topic describes this demo, which models the theory behind signal transmission:
Suppose that you want to create a model that more accurately portrays what happens during cellular phone communication. A better approximation of a real-world system would involve the quantization of the residual and reflection coefficients before they are transmitted. For information on how to design a scalar quantizer to accomplish such a task, see Creating a Scalar Quantizer.
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