{"title":"Instrumental Estimation of E-model Equipment Impairment Factor Parameters for Super-wideband Opus Codec","authors":"Mohannad Al-Ahmadi, P. Počta, H. Melvin","doi":"10.1109/ISSC.2019.8904932","DOIUrl":null,"url":null,"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.","PeriodicalId":312808,"journal":{"name":"2019 30th Irish Signals and Systems Conference (ISSC)","volume":"132 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2019-06-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"2","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"2019 30th Irish Signals and Systems Conference (ISSC)","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/ISSC.2019.8904932","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
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.