Auditory Model Based 2.4kbps Speech CODEC and Its Real-Time Implementation Using Single TMS320C5410 DSP Chip
In this paper, both sinusoidal representation and secondorder difference cochlear model are used to extract speech feature. Linear interpolation and non-linear smoothing are both used to raise the quality of synthesized speech. A 2.4kbps low bit-rate speech CODEC algorithm is developed after using vector quantisation (VQ) technology to quantise speech features. A single TMS320C5410 DSP chip is adopted to realize a real-time 2.4kbps auditory model based speech CODEC. The experimental results show that the synthesized speech has MOS3.8 quality.
Xiaoqing Yu Wanggen Wan Lei Ma Ning Wang Daniel P.K. Lun
Multimedia Innovation Centre, The Hong Kong Polytechnic University, Kowloon, Hong Kong Centre for Multimedia Signal Processing, Department of Electronic and Information Engineering The Ho
国际会议
8th International Conference on Neural Information Processing(ICONIP 2001)(第八届国际神经信息处理大会)
上海
英文
1509-1514
2001-11-14(万方平台首次上网日期,不代表论文的发表时间)