Chrome Version: M64

In the active experiment to launch WebRTC's fix for long delays when using Opus with DTX/CNG (https://bugs.chromium.org/p/chromium/issues/detail?id=780849), we have seen in Hangouts stats that the target audio jitter buffer level is increasing with the experiment. This is unexpected, and cannot be explained. To get a second opinion, we would like to add the same metric to UMA (where the population is different).

The UMA stat was added to WebRTC in https://webrtc-review.googlesource.com/c/src/+/26740, but that did not make the M64 WebRTC cut. Since the experiment will roll to M64 Beta, I would like to have this stat available for the Beta population. Hence, requesting a merge.