摘要
文中提出了一种新的适用于实时多媒体应用领域的音频编码算法．该算法首先对音频信号进行小波包分解，然后在小波域中计算掩蔽阈值，最后根据从心理声学模型得到的信号一掩蔽比来对各子带小波系数进行动态比特分配、量化和编码．实验结果表明该算法将 ＣＤ音频信号压缩到 ６４ Ｋｂｐｓ时，恢复信号的分段信噪比为 ３２．３２ｄＢ，主观上感觉无失真．该算法计算简单，可在不需任何附加硬件的 Ｐｅｎｔｉｕｍ １３３ ＭＨｚ个人计算机上实现实时音频编码．
A novel audio coding algorithm suitable for real time multimedia applications is presented in this paper. Firstly, audio signals are decomposed based on wavelet packet using this algorithm, and then the masking threshold is calculated in wavelet domain. Finally, according to the signal-mask ratio obtained with psychoacoustic model, dynamic bits allocation, quantization, and encoding to the wavelet coefficients of each subband are performed. The experimental results indicate that when compact disc audio is compressed to 64 Kbps utilizing this algorithm, the segmentation signal-noise ratio of the recovery signal is 32.32 dB and with virtually no perceptual degradation in sound quality. Computation of this algorithm is simple and real time encoding can be performed in the Pentium 133 MHz computer with no resort to an expensive and dedicated hardware.
出处
《计算机研究与发展》
EI
CSCD
北大核心
2000年第3期329-335,共7页
Journal of Computer Research and Development
基金
国家"八六三"高技术研究发展计划项目!项目编号863-306-ZT03-01-2
关键词
音频编码
小波包
心理声学模型
多媒体
算法
audio coding, wavelet packet, psychoacoustic model, dynamic bits allocation
