adaptivePtime property for RTCRtpEncodingParameters

Web RTC

Adds the adaptivePtime property to RTCRtpEncodingParameters(https://w3c.github.io/webrtc-pc/#rtcrtpencodingparameters). This new flag sets a sender in a real-time communication (RTC) system to enable/disable adaptive packet rate.

As the packet rate is a big determining factor to the overall bitrate of an audio stream, an optimal congestion control should be allowed to adapt the packet rate. The audio packet rate is analogous to the video frame rate, which also plays an important role in the video bitrate adaptation. Although adaptive packet rate may be ubiquitously beneficial, we need this API for applications to enable and disable it, since, otherwise, it may introduce interoperability problems. Some implementations have taken a fixed packet rate as an assumption, and thus may fail or perform sub-optimally with an adaptive packet rate. Therefore it is needed to add a flag for applications to enable or disable the adaptive packet rate.

Documentation

Specification

Established standard

Status in Chromium

Blink>WebRTC


Origin trial (tracking bug) in:

  • Chrome for desktop release 85
  • Chrome for Android release 85
  • Android WebView release 85

Consensus & Standardization

After a feature ships in Chrome, the values listed here are not guaranteed to be up to date.

  • No signal
  • No signal
  • No signal
  • Positive

Owners

Intent to Prototype url

Intent to Prototype thread

Last updated on 2020-08-10