Abstract

We proposed a new non-uniform sampling and quantization using variables low-pass filter with correlation. It focuses on the naturalness and intelligibility of speech synthesis applications and the compression and signal-to-noise ratio of speech transmission applications. However, it is well known that when conventional sampling methods are applied directly to speech signal, the required amount of data is comparable to or more than that of uniform sampling method. To overcome this problem, a new non-uniform methods is proposed, in which time domain coding is applied to two low-pass filters in lower bandwidth and the remain signals are compensated by the Gaussian white signal, which is used to get high quality speech by correlation of signal .

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