用于Voip系统的自适应音频量化器

Ricardo Bertagna, R. Mello, L. Yang
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引用次数: 0

摘要

互联网的发展要求开发新的技术来支持实时的多媒体传输,如图像、数据库访问、音频和视频。这种发展需要新的服务和支持,如IP语音(VoIP),其主要动机是低成本的通信和管理。VoIP系统推动了这项工作,提出了一种名为IQ(区间量化器)的自适应音频量化器来降低数据维数,从而降低熵,从而实现更好的音频压缩。这个量化器是自适应的,因为它有一个容错参数,可以根据可用的网络带宽变化,允许适应通信。在传输后,通过在Z平面上使用具有复杂极点的滤波器来改善音频。这个滤波器衰减不重要的频率,使敏感的频率对人的听觉有特权。结果证实IQ和过滤器提供了良好的质量(使用平均意见得分指标测量)和压缩比。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
An Adaptive Audio Quantizer for Voip Systems
The Internet evolution has been requiring the development of new technology to support multimedia transmission such as images, database access, audio and video in realtime. Such development needs new services and supports like the voice over IP (VoIP) which has a main motivation in the low cost communication and management. VoIP systems have motivated this work which proposes an adaptive audio quantizer named IQ (intervalar quantizer) to reduce the data dimensionality and consequently the entropy, what allows better audio compression. This quantizer is adaptive because it has an error tolerance parameter which can be varied according to the available network bandwidth, allowing to adapt communication. After transmitting, the audio is improved by using a filter with complex poles in the Z plan. This filter attenuates non-important frequencies, privileging the sensitive ones to human audition. Results confirm that IQ and the filter offer good quality (measured using the mean opinion score metrics) and compress ratio.
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