摘要
音频编码要解决的问题是以最小感知失真用低速率表达音频信号.本文设计了一种基于正交小波变换和音质模型的自适应比特分配音频编码算法,它可以将 1411.2kbit/s的双声道立体声高保真音频信号压缩成低至32kbit/s的速率,并保持很好的音频质量.
The Problem of audio coding is to achieve a low bit rate in the digital representation of an actio signal with minimum perceived loss of quality, A Hi-Fi audio coding algorithm based on orthogonal Wavelet transform, psychoacoustic model and adaptive bit allocation is designed. The 1411.2kbit/s stereo Hi-Fi audio signals can be compressed to the bit late as low as 32kbit/s with good audio quality.
出处
《电子学报》
EI
CAS
CSCD
北大核心
2000年第1期26-29,共4页
Acta Electronica Sinica
基金
国家自然科学基金
综合业务网理论及关键技术国家重点实验室资助课题
关键词
音频编码
小波变换
音质模型
信源编码
audio coding
wavelet transform
psychoacoustic model
source coding