)]}'
{
  "commit": "70ffead25675c4761467755f9be844335dd59dba",
  "tree": "babc739674a6b2b6a47271251dc87f7f1bac9ebb",
  "parents": [
    "f39f7d931c11045fe0ba842e4eba9b816f0288ca"
  ],
  "author": {
    "name": "Danil Chapovalov",
    "email": "danilchap@webrtc.org",
    "time": "Wed Jul 20 13:26:59 2016"
  },
  "committer": {
    "name": "Danil Chapovalov",
    "email": "danilchap@webrtc.org",
    "time": "Wed Jul 20 13:27:09 2016"
  },
  "message": "Reimplemented fix for bogus RTP timestamp in RTCP packet created before RTP packet.\n\nNow it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too.\n\nIt also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet.\n\nBUG\u003dwebrtc:1600\nR\u003dpbos@webrtc.org, stefan@webrtc.org\n\nReview URL: https://codereview.webrtc.org/1639253007 .\n\nCr-Commit-Position: refs/heads/master@{#13503}\n",
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