Nowadays, widely used videoconferencing software has been diffused even further by the social distancing measures adopted during the SARS-CoV-2 pandemic. However, none of the Web-based solutions currently available support high-fidelity stereo audio streaming, which is a fundamental prerequisite for networked music applications. This is mainly because of the fact that the WebRTC RTCPeerConnection standard or Web-based audio streaming do not handle uncompressed audio formats. To overcome that limitation, an implementation of 16-bit pulse code modulation (PCM) stereo audio transmission on top of the WebRTC RTCDataChannel, leveraging Web Audio and AudioWorklets, is discussed. Results obtained with multiple configurations, browsers, and operating systems showthat the proposed approach outperforms theWebRTC RTCPeerConnection standard in terms of audio quality and latency, which in the authors' best case to date has been reduced to only 40 ms between twoMacBooks on a local area network.

Authors: Sacchetto, Matteo; Gastaldi, Paolo; Chafe, Chris; Rottondi, Cristina; Servetti, Antonio
Affiliations: Department of Electronics and Telecommunications, Politecnico di Torino, Italy; Department of Energy, Politecnico di Torino, Italy; Center for Computer Research in Music and Acoustic, Stanford University, CA; Department of Electronics and Telecommunications, Politecnico di Torino, Italy; Department of Control and Computer Engineering, Politecnico di Torino, Italy(See document for exact affiliation information.)
JAES Volume 70 Issue 11 pp. 926-937; November 2022
Publication Date: November 15, 2022 Import into BibTeX
Permalink: https://www.aes.org/e-lib/browse.cfm?elib=22017

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DOI: https://doi.org/10.17743/jaes.2022.0021

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