Derivation of E-model Equipment Impairment Factors for Narrowband and Wideband Opus Codec Using the Instrumental Method

Mohannad Al-Ahmadi, P. Počta, H. Melvin
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Abstract

Real-time multimedia applications like Web realtime communication WebRTC support a wide range of codecs, from the standard narrowband up to fullband codecs. The IETF standardized Opus codec is the default codec utilized by WebRTC speech and audio applications, by supporting a wide range of bitrates. In current best effort networks, network impairments such as packet loss, delay and jitter affect the quality of VoIP. To assess the impact of such impairments in order to estimate the quality experienced by the end users of speech applications, the E-model standardized in ITU-T Rec. G.107 can be used. In this paper we derive codec-specific parameters required by the E-model to estimate the quality degradation in speech applications deploying narrowband and wideband Opus codec, namely the equipment impairment factor Ie and packet loss robustness factor Bpl. We followed the ITU-T methods designed for this purpose and share the results arising from all the experiments covering all the narrowband and wideband Opus codec conditions. 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.
用仪器法推导窄带和宽带工作码编解码器的e型设备损伤因子
像Web实时通信这样的实时多媒体应用支持广泛的编解码器,从标准窄带到全带编解码器。IETF标准化的Opus编解码器是WebRTC语音和音频应用程序使用的默认编解码器,支持广泛的比特率。在目前的最佳努力网络中,诸如丢包、延迟和抖动等网络缺陷会影响VoIP的质量。为了评估这种损害的影响,以估计语音应用的最终用户所体验到的质量,可以使用ITU-T Rec. G.107标准的e模型。在本文中,我们推导了e模型所需的编解码器特定参数,以估计部署窄带和宽带Opus编解码器的语音应用中的质量退化,即设备损伤因子Ie和丢包鲁棒性因子Bpl。我们遵循为此目的而设计的ITU-T方法,并分享涵盖所有窄带和宽带Opus编解码条件的所有实验结果。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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