高保真多声道音频编码与Karhunen-Loeve变换

Dai Yang, H. Ai, C. Kyriakakis, C.-C. Jay Kuo
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引用次数: 34

摘要

提出了一种基于MPEG-2高级音频编码(AAC)的高保真多通道音频压缩算法。在预处理阶段对多声道音频信号应用Karhunen-Loeve变换(KLT)去除道间冗余。然后,用改进的AAC主剖面编码器对去相关信道中的信号进行压缩。最后,采用通道传输控制机制对比特流进行重新组织,使多通道音频比特流在异构网络上传输时具有质量可扩展性。实验结果表明,在64 kbit/sec/ch的常规比特率下,与AAC相比,该算法在保持相似计算复杂度的情况下取得了更好的性能。当比特流以较低的比特率传输到窄带终端用户时,某些通道中的数据包可能会被丢弃,并且稍微降级,但是全通道音频仍然可以以合理的方式重建,而不需要任何额外的计算成本。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
High-fidelity multichannel audio coding with Karhunen-Loeve transform
A new quality-scalable high-fidelity multichannel audio compression algorithm based on MPEG-2 advanced audio coding (AAC) is presented. The Karhunen-Loeve transform (KLT) is applied to multichannel audio signals in the preprocessing stage to remove interchannel redundancy. Then, signals in decorrelated channels are compressed by a modified AAC main profile encoder. Finally, a channel transmission control mechanism is used to re-organize the bitstream so that the multichannel audio bitstream has a quality scalable property when it is transmitted over a heterogeneous network. Experimental results show that, compared with AAC, the proposed algorithm achieves a better performance while maintaining a similar computational complexity at the regular bit rate of 64 kbit/sec/ch. When the bitstream is transmitted to narrowband end users at a lower bit rate, packets in some channels can be dropped, and slightly degraded, yet full-channel, audio can still be reconstructed in a reasonable fashion without any additional computational cost.
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