Modelling Speech Quality for NB and WB SILK Codec for VoIP Applications

M. Goudarzi, Lingfen Sun, E. Ifeachor
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引用次数: 14

Abstract

In the last decade, VoIP telephony has gained a tremendous popularity. Skype is one of the most successful and popular VoIP services which has inspired a new generation of VoIP and multimedia users. SILK speech codec is the latest development by Skype and has been integrated into the current version of Skype and is expected to be incorporated into new and emerging mobile devices such as iphone and soft phones. One of the major challenges in every VoIP service is to find an easily accessible objective quality model to predict/measure the perceived speech quality or the degree of user satisfaction. In this paper, we present a regression-based model to quantify the speech quality of the wideband (WB) and narrowband (NB) SILK codec for VoIP applications. The developed model uses the network level parameter (i.e., packet loss) and the application level parameter (i.e., send bit rate) to predict the perceived voice quality in terms of the Mean Opinion Score (MOS). Subjective tests were also carried out to validate the model and good accuracy was achieved (97% for wideband and 91% for narrowband). The developed model can be easily implemented in soft phones or mobile devices to predict voice quality for SILK codec in VoIP applications and can also be used for real-time adaptation and control of VoIP applications to further explore the adaptive feature of the SILK codec in future mobile devices or softphones.
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用于VoIP应用的NB和WB SILK编解码器的语音质量建模
在过去的十年中,VoIP电话获得了极大的普及。Skype是最成功和最受欢迎的VoIP服务之一,它激发了新一代VoIP和多媒体用户。SILK语音编解码器是Skype的最新开发,已集成到当前版本的Skype中,并有望集成到新的和新兴的移动设备中,如iphone和软电话。每个VoIP服务的主要挑战之一是找到一个易于访问的客观质量模型来预测/测量感知语音质量或用户满意度。在本文中,我们提出了一个基于回归的模型来量化用于VoIP应用的宽带(WB)和窄带(NB) SILK编解码器的语音质量。所开发的模型使用网络级别参数(即丢包)和应用级别参数(即发送比特率)来根据平均意见得分(Mean Opinion Score, MOS)预测感知语音质量。还进行了主观测试来验证模型,并取得了良好的准确性(宽带为97%,窄带为91%)。该模型可方便地在软电话或移动设备中实现,用于预测VoIP应用中SILK编解码器的语音质量,也可用于VoIP应用的实时适应和控制,进一步探索未来移动设备或软电话中SILK编解码器的自适应特性。
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