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A bandwidth extension method based on audio classification was proposed.Time series of audio signals were classified into four types based on recurrence plot and recurrence quantification analysis,and the fine spectrums were recovered by taking advantage of four methods respectively.In addition,the spectrum envelope and energy gain were adjusted by Gaussian mixture model and codebook mapping on the basis of soft decision respectively.Subjective and objective testing results indicate that the proposed method has good quality compared with conventional blind bandwidth extension methods,and the performance of ITU-T G.722.1 codec with the proposed algorithm is better than that of G.722.1C codec at the same bit rate.

A bandwidth extension method based on audio classification was proposed.Time series of audio signals were classified into four types based on recurrence plot and recurrence quantification analysis,and the fine spectrums were recovered by taking advantage of four methods respectively.In addition,the spectrum envelope and energy gain were adjusted by Gaussian mixture model and codebook mapping on the basis of soft decision respectively.Subjective and objective testing results indicate that the proposed method has good quality compared with conventional blind bandwidth extension methods,and the performance of ITU-T G.722.1 codec with the proposed algorithm is better than that of G.722.1C codec at the same bit rate.

Based on the international telecommunication union telecommunication standardization sector （ITU-T） recommendation G.729.1 and modified modulated lapped transform （MLT） coding, a super-wideband embedded variable bit-rate speech and audio coding algorithm was proposed, the bit-rates of this codec was from 8kbit/s to 64kbit/s. The information in the frequencies of 0~7 kHz was encoded by G.729.1 codec at 8~32kbit/s, the information in the frequencies of 7~14 kHz was encoded by the transform coding at 36、40 and 48kbit/s, and G.729.1 residual signal’s MDCT （modified discrete cosine transform） was encoded by the transform coding at 56 and 64kbit/s. The objective and subjective listening tests show that this codec has good performance compared with Terms of Reference given by ITU-T.

Based on the international telecommunication union telecommunication standardization sector （ITU-T） recommendation G.729.1 and modified modulated lapped transform （MLT） coding, a super-wideband embedded variable bit-rate speech and audio coding algorithm was proposed, the bit-rates of this codec was from 8kbit/s to 64kbit/s. The information in the frequencies of 0~7 kHz was encoded by G.729.1 codec at 8~32kbit/s, the information in the frequencies of 7~14 kHz was encoded by the transform coding at 36、40 and 48kbit/s, and G.729.1 residual signal’s MDCT （modified discrete cosine transform） was encoded by the transform coding at 56 and 64kbit/s. The objective and subjective listening tests show that this codec has good performance compared with Terms of Reference given by ITU-T.

The psychoacoustic models used in perceptnal audio coding are described. The algorithm and realtime hardware implement complexity of MPEG/Audio,Dolby AC-2 ,and AC-3 are introduced.The result of the MPEG/Audio multichannel subject listening tests are introduced also.

The psychoacoustic models used in perceptnal audio coding are described. The algorithm and realtime hardware implement complexity of MPEG/Audio,Dolby AC-2 ,and AC-3 are introduced.The result of the MPEG/Audio multichannel subject listening tests are introduced also.