)]}'
{
  "commit": "883eefc59e96da2dcb23a36f178610a92587cd73",
  "tree": "ddc6f6b179878a5ec2fc5efc4566df64c651eff1",
  "parents": [
    "6e436d1cc0d54ed39bc28d4dac2e9439de9a5b65"
  ],
  "author": {
    "name": "Henrik Boström",
    "email": "hbos@webrtc.org",
    "time": "Mon May 27 11:40:25 2019"
  },
  "committer": {
    "name": "Commit Bot",
    "email": "commit-bot@chromium.org",
    "time": "Mon May 27 12:45:22 2019"
  },
  "message": "Implement RTCRemoteInboundRtpStreamStats for both audio and video.\n\nThis implements the essentials of RTCRemoteInboundRtpStreamStats. This\nincludes:\n- ssrc\n- transportId\n- codecId\n- packetsLost\n- jitter\n- localId\n- roundTripTime\nhttps://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*\n\nThe following members are not implemented because they require more\nwork...\n- From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded,\n  packetsRepaired, burstPacketsLost, burstPacketsDiscarded,\n  burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate,\n  gapLossRate and gapDiscardRate.\n- From RTCRemoteInboundRtpStreamStats: fractionLost.\n\nBug: webrtc:10455, webrtc:10456\nChange-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d\nReviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067\nCommit-Queue: Henrik Boström \u003chbos@webrtc.org\u003e\nReviewed-by: Harald Alvestrand \u003chta@webrtc.org\u003e\nCr-Commit-Position: refs/heads/master@{#28073}\n",
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