Create an origin trial that when enabled allows users to change the size of the WebRTC jitter buffer and read a new statistic counting the number of times the buffer is flushed. This will be allow us to explore the performance impact of tweaking this value. If it proves useful we'll either tweak the WebRTC implementation or start working to add it the standardized WebRTC APIs.
Docs: https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit