[webrtc] Adds UMA histogram descriptions and updates histograms at call end

[webrtc] Adds UMA histogram descriptions and updates histograms at call end

WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps and WebRTC.Audio.NumOfAecSystemDelayJumps now have descriptions added.
These histograms need to be updated at the end of a webrtc call.
We can not internally in webrtc::AudioProcessing rely on the destructor, since the object is not necessarily destroyed.

The underlying WebRTC change at https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+/c1439edde708296bfe909fcb9521dfe059ad44d8
rolled into chromium at https://crrev.com/f541436aa4cf51fb2ade83127b230cdc4a69c72b

Tested with apprtc in chromium ToT where three jumps (200, 50 and 60 ms) were triggered.
The 60 and 200 ms jumps were caught by the metrics as expected.

BUG=488124

Review URL: https://codereview.chromium.org/1216103004

Cr-Commit-Position: refs/heads/master@{#337639}