基于分形音频编码和误差补偿的高合成音频压缩模型

Q2 Computer Science
A. Ali, Loay E. George
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引用次数: 1

摘要

本研究提出了一个使用分形编码提高音频文件质量的模型,特别是当需要高压缩比时。提出了基于传统分形编码和提升小波变换的高合成音频压缩模型(HSACM)。讨论了各种提升小波变换族和层次,并讨论了它们对音频重构文件的影响。采用GTZAN数据集的音频文件和数据压缩的标准测量值来评估所提出的模型。结果表明,在块长为50的最坏情况下,采用3级和2级提升小波变换,PSNR分别从34.1提高到44.8 dB和34.1提高到40.5 dB。因此,PSNR分别提高了10 dB和5 dB,压缩比分别降低了6.2和12.5%。此外,可以注意到,采用Haar基的提升小波变换,db1、db4、db5、cdf1.1和cdf2.2的音频质量较高,而db6、db8、sym7和sym8的音频质量最差。此外,将HSACM的性能与现有工作进行了比较,以突出其性能。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
High Synthetic Audio Compression Model Based on Fractal Audio Coding and Error-Compensation
This study presented a model for improving audio files quality using fractal coding specifically when a high compression ratio is required. The proposed high synthetic audio compression model which can be called (HSACM) is based on conventional fractal coding and lifting wavelet transform. Various lifting wavelet transform families and levels are used and their effects on the reconstructed audio files are discussed as well. Audio files from GTZAN dataset and standard measurements for data compression are used in the evaluation of the proposed model. The results reveal that using block length 50 samples which is the worst case, PSNR is increased, on average, from 34.1 to 44.8 dB and from 34.1 to 40.5 dB using lifting wavelet transform with 3 and 2 levels, respectively. Thus, the PSNR is improved by 10 and 5 dB with slightly reducing the compression ratio by 6.2 and 12.5%, respectively. Moreover, it can be noticed that adopting lifting wavelet transform with basis Haar, db1, db4, db5, cdf1.1 and cdf2.2 provide higher audio quality while db6, db8, sym7 and sym8 give the worst audio quality. Furthermore, the performance of HSACM is compared with that of existing work to highlight its performance.
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来源期刊
Annals of Emerging Technologies in Computing
Annals of Emerging Technologies in Computing Computer Science-Computer Science (all)
CiteScore
3.50
自引率
0.00%
发文量
26
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