Abstract
To improve the quality and intelligibility of telephone-band speech coding, a new time–frequency representation based on a tight framelet packet transform is proposed in this paper. In the context of speech coding, the effectiveness of this representation stems from its resilience to quantization noise, and reconstruction stability. Moreover, it offers a sub-band decomposition and good time–frequency localization according to the critical bands of the human ear. The coded signal is obtained using dynamic bit allocation and optimal quantization of normalized framelet coefficients. The performances of the corresponding method are compared to the critically sampled wavelet packet transform. Extensive simulation revealed that the proposed speech coding scheme, which incorporates the tight framelet packet transform performs better than that based on the critically sampled wavelet packet transform. Furthermore, it ensures a high bit-rate reduction with negligible degradation in speech quality. The proposed coder is found to outperform the standard telephone-band speech coders in term of objective measures and subjective evaluations including a formal listening test. The subjective quality of our codec at 4 kbps is almost identical to the reference G.711 codec operating at 64 kbps.
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