Web-Based Networked Music Performances via WebRTC: A Low-Latency PCM Audio Solution

Sacchetto M, Gastaldi P, Chafe C, Rottondi C, Servetti A. Web-Based Networked Music Performances via WebRTC: A Low-Latency PCM Audio Solution. J Audio Eng Soc. 2022;70(11):926-937. doi:10.17743/jaes.2022.0021

@article{Sacchetto2022_22017,
  author = {Sacchetto, Matteo and Gastaldi, Paolo and Chafe, Chris and Rottondi, Cristina and Servetti, Antonio},
  title = {{Web-Based Networked Music Performances via WebRTC: A Low-Latency PCM Audio Solution}},
  journal = {Journal of the Audio Engineering Society},
  volume = {70},
  number = {11},
  pages = {926--937},
  year = {2022},
  month = nov,
  publisher = {Audio Engineering Society},
  doi = {10.17743/jaes.2022.0021},
  url = {https://doi.org/10.17743/jaes.2022.0021}
}

TY  - JOUR
TI  - Web-Based Networked Music Performances via WebRTC: A Low-Latency PCM Audio Solution
AU  - Sacchetto, Matteo
AU  - Gastaldi, Paolo
AU  - Chafe, Chris
AU  - Rottondi, Cristina
AU  - Servetti, Antonio
T2  - Journal of the Audio Engineering Society
J2  - J. Audio Eng. Soc.
VL  - 70
IS  - 11
SP  - 926
EP  - 937
PY  - 2022
DA  - 2022/11/06
DO  - 10.17743/jaes.2022.0021
UR  - https://doi.org/10.17743/jaes.2022.0021
PB  - Audio Engineering Society
LA  - en
AB  - 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.
ER  -