Instrumental Estimation of E-model Equipment Impairment Factor Parameters for Super-wideband Opus Codec

Mohannad Al-Ahmadi, P. Počta, H. Melvin
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引用次数: 2

Abstract

WebRTC includes a set of novel technologies and standards that provide high quality audio, video and arbitrary data exchange. By enabling rich interactivity experience, We-bRTC allows the users to directly interact with use of web and mobile applications without the need for additional plugins. The open standard Opus codec is a default audio codec employed by WebRTC for speech communication and music streaming with a wide range of bitrates supported. This range of bitrates is available for narrowband (300–3400 Hz), wideband (100–7000 Hz) up to super-wideband (50–14000 Hz) audio bandwidths. To estimate a quality experienced by the end user for voice transmission service, E-model standardized in the ITU-T Rec. G.107 (a narrowband version) and the ITU-T Rec. G.107.1 (a wideband version) can be used. With the upcoming extension towards the super-wideband E-model, it will be soon possible to estimate the quality of narrowband, wideband as well as super-wideband speech communication. In this paper, we calculate two codec specific parameters, namely the equipment impairment factor Ie and packet loss robustness Bpl for the super-wideband mode of the Opus codec using the instrumental method. The derived values make it possible to integrate the E-model in realtime communication applications including WebRTC to assess the quality experienced by the end user.
超宽带Opus编解码器e型设备损伤因子参数的仪器估计
WebRTC包括一套新颖的技术和标准,提供高质量的音频、视频和任意数据交换。通过丰富的交互体验,We-bRTC允许用户直接与使用web和移动应用程序进行交互,而无需额外的插件。开放标准Opus编解码器是WebRTC用于语音通信和音乐流的默认音频编解码器,支持广泛的比特率。此比特率范围可用于窄带(300-3400 Hz),宽带(100-7000 Hz)到超宽带(50-14000 Hz)音频带宽。为了评估最终用户体验到的语音传输服务质量,可以使用ITU-T Rec. G.107(窄带版本)和ITU-T Rec. G.107.1(宽带版本)标准化的E-model。随着即将向超宽带e型扩展,很快就可以对窄带、宽带以及超宽带语音通信的质量进行评估。本文采用仪器方法计算了Opus编解码器超宽带模式下的设备损伤因子Ie和丢包鲁棒性Bpl两个编解码器特定参数。得到的值使得将e -模型集成到实时通信应用(包括WebRTC)中以评估最终用户体验的质量成为可能。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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