| 摘要 |
In a speech encoder, high-frequency components of input digital speech samples are emphasized by a preemphasis filter (11). From the preemphasized samples a spectral parameter (ai) is derived at frame intervals. The input digital samples are weighted by a weighting filter (13) according to a characteristic that is inverse to the characteristic of the preemphasis filter (11) and is a function of the spectral parameter (aj). A codebook (18, 19) is searched for an optimum fricative value in response to a pitch parameter that is derived by an adaptive codebook (16) from a previous fricative value (v(n)) and a difference between the weighted speech samples and synthesized speech samples which are, in turn, derived from past pitch parameters and optimum fricative values, whereby the difference is reduced to a minimum. Index signals representing the spectral parameter, pitch parameter and optimum fricative value are multiplexed into a single data stream.
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