基于差别化衰减的混音信号溢出自适应处理算法

姜健

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基于差别化衰减的混音信号溢出自适应处理算法

自动化与计算机技术 | 更新时间：2026-09-14

- 基于差别化衰减的混音信号溢出自适应处理算法
- An Adaptive Algorithm for Audio Mixing Overflow Base on Differentiated Attenuation
- 中北大学学报（自然科学版） 2012年33卷第4期页码：431-436
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  部队,63961
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- 纸质出版：2012
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姜健. 基于差别化衰减的混音信号溢出自适应处理算法[J]. 中北大学学报（自然科学版）, 2012,33(4):431-436.

姜健. An Adaptive Algorithm for Audio Mixing Overflow Base on Differentiated Attenuation[J]. 2012, 33(4): 431-436.
姜健. 基于差别化衰减的混音信号溢出自适应处理算法[J]. 中北大学学报（自然科学版）, 2012,33(4):431-436. DOI：

姜健. An Adaptive Algorithm for Audio Mixing Overflow Base on Differentiated Attenuation[J]. 2012, 33(4): 431-436. DOI：

摘要

针对目前视频会议系统中运用自适应算法解决混音信号溢出问题中出现的弱信号被进一步衰减的问题

提出了一种改进算法.该算法采用在信号幅值上设置衰减阀值参量

再将语音信号各采样点幅值与此阀值对比大小

以决定是否对其衰减的方法

差别化混音信号的衰减过程

达到既消除溢出噪声又保持整体音质的目的.通过利用已采集好的4路语音信号在桌面计算机平台混音测试

分析了诸算法的实验效果.结果表明

经该算法处理的语音信号提高了信噪比

保持了弱采样点信号声强

适于音质要求清晰、连贯的高级别视频会议.

Abstract

To overcome the overflow and the attenuation of weak audio mixing signals by using adaptive algorithm in video conference system

a new method was proposed based on the adaptive algorithm.An attenuation threshold was set on the signal amplitude

with whom the size of sample amplitude was compared to determine whether attenuation to be adopted.The proposed attenuation process could eliminate the overflow noise and guaranteed the overall acoustic quality.Audio mixing tests were implemented by analyzing each algorithm′s effect of four prepared speech signals on the platform of desktop PC.The results showed that the speech signal′s SNR was improved which was treated with the proposed algorithm.It holds the weak sample′s sound intensity

suits to classical video conference system with higher demand of clear and coherent in speech.

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references

蔡必强.视频会议中混音技术研究[J].现代电子技术,2006,235(20):85-87.Cai Biqiang.Research on audio mixing technology invideo conference[J].Modern Electronics Technique,2006,235(20):85-87.(in Chinese)

邹力子,刘林.改进的视频会议系统混音算法及实现[J].通信技术,2011,44(2):120-122.Zou Lizi,Liu Lin.An improved audio mixingalgorithm for video conferencing system and itsimplementation[J].Communications Technology,2011,44(2):120-122.(in Chinese)

Gonazález A J,Hussein A W.Audio mixing forinteractive multimedia communications[C].//JCIS'98.North Carolina:Binary Information Press,1998:217-220.

张海峰,白骋宇.多媒体会议系统音频多点处理器的软件设计[J].机电工程,2010,27(6):104-107.Zhang Haifeng,Bai Chengyu.Software design ofaudio multi-point processor of multimedia conferencesystem[J].Journal of Mechanical&ElectricalEngineering,2010,27(6):104-107.(in Chinese)

王文林,廖建新,朱晓民,等.多媒体会议中新型快速实时混音算法[J].电子与信息学报,2007,29(3):690-695.Wang Wenlin,Liao Jianxin,Zhu Xiaomin,et al.Anovel fast real-time audio mixing algorithm inmultimedia conference[J].Journal of Electronics&Information Technology,2007,29(3):690-695.(inChinese)

樊星,顾伟康,叶秀清.多媒体会议中的快速实时自适应混音方案研究[J].软件学报,2005,16(1):108-115.Fan Xing,Gu Weikang,Ye Xiuqing.Fast real-timeadaptive audio mixing schemes in multimediaconferencing[J].Journal of Software,2005,16(1):108-115.(in Chinese)

韩钰,普杰信.一种新的网络电话会议混音算法[J].计算机应用,2010,30(2):564-566.Han Yu,Pu Jiexin.New mixing algorithm fornetwork and telephone conference[J].Journal ofComputer Applications,2010,30(2):564-566.(inChinese)

马旋,王衡,汪国平,等.视频会议中混音后溢出问题的研究及解决方法[C].//第13届全国多媒体学术会议(NCMT2004)论文集.宁波,2004.

周敬利,马志龙,范晔斌,等.一种新的多媒体会议实时混音方案[J].小型微型计算机系统,2009,30(1):169-172.Zhou Jingli,Ma Zhilong,Fan Yebin,et al.Real-timeaudio mixing arithmetic in multipoint conferencing[J].Journal of Chinese Computer Systems,2009,30(1):169-172.(in Chinese)

毕厚杰,陈启美,方晖.IP宽带通信网络技术[M].北京:北京邮电大学出版社,2004.

徐保民,王秀玲.一个改进的混音算法[J].电子与信息学报,2003,25(12):1709-1713.Xu Baomin,Wang Xiuling.An improved mixedaudio algorithm[J].Journal of Electronics&Information Technology,2003,25(12):1709-1713.(in Chinese)

刘,桂志国.一种基于模糊集理论的贝叶斯PET重建算法[J].中北大学学报(自然科学版),2011,32(5):630-635.Liu Yi,Gui Zhiguo.A bayesian algorithm based onfuzzy theory for PET reconstruction[J].Journal ofNorth University of China(Natural ScienceEdition),2011,32(5):630-635.(in Chinese)

Cherdyntsev E S.Multimedia networks[M].Tomsk:Tomsk Polytechnic University PublishingHouse,2011.

Zou Jianfeng,Shen Wei.A moving-window basedaudio mixing algorithm[C].//CSIE 2009.LosAngeles/Anaheim:IEEE Computer Society,2009:248-251.

ITU-T Recommendation G.711.Pulse CodeModulation(PCM)of Voice Frequencies[S].Geneva:Telecommunication Standardization Sectorof ITU,1988,1993.

李文海,毛京丽,石方文.数字通信原理[M].北京:人民邮电出版社,2001.

张微.VOIP电话会议系统中混音模块关键技术的研究与实现[D].上海:华东师范大学,2007.

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