| 摘要 |
A method and device for improving the quality of speech signals transmitted using an audio bandwidth between 300 Hz and 3.4 kHz. After the received speech signal is divided into frames, zeros are inserted between samples to double the sampling frequency. The level of these aliased frequency components is adjusted using an adaptive algorithm based on the classification of the speech frame. Sound can be classified into sibilants and non-sibilants, and a non-sibilant sound can be further classified into a voiced sound and a stop consonant. The adjustment is based on parameters, such as the number of zero-crossings and energy distribution, computed from the spectrum of the up-sampled speech signal between 300 Hz and 3.4kHz. A new sound with a bandwidth between 300 Hz and 7.7kHz is obtained by inverse Fourier transforming the spectrum of the adjusted, up-sampled sound.
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