)]}'
{
  "commit": "fcc398163304ba0c26df5ff270905e349d3accc3",
  "tree": "b3eb9c573cc31aecf9b390267bd2ad980d37dd93",
  "parents": [
    "992a868393f322593e69d8a44d79dcb8c2f32a46"
  ],
  "author": {
    "name": "Ilya Nikolaevskiy",
    "email": "ilnik@webrtc.org",
    "time": "Tue Oct 30 15:47:48 2018"
  },
  "committer": {
    "name": "Commit Bot",
    "email": "commit-bot@chromium.org",
    "time": "Tue Oct 30 15:47:59 2018"
  },
  "message": "Revert \"Use only first payload timestamp for RTCP SR generation for audio\"\n\nThis reverts commit 9a0662ac7e4a3bc6b3a316397a7fdf25f0025d35.\n\nReason for revert: breaks some av sync perf tests\n\nOriginal change\u0027s description:\n\u003e Use only first payload timestamp for RTCP SR generation for audio\n\u003e \n\u003e Since now RTP rate is set correctly for audio, there\u0027s no need to\n\u003e use the very last data packet rtp/capture timestamps for generating\n\u003e RTCP SR packets.\n\u003e \n\u003e Using only one (first) packet timestamp eliminates the jitter between\n\u003e rtp and capture timestamps for audio. This jitter comes from the fact\n\u003e that capture timestamp for audio is unknown and we generate bogus\n\u003e timestamp at arbitrary, non-constant offset from the real capture time.\n\u003e \n\u003e Bug: webrtc:9905\n\u003e Change-Id: I855556184cfe994be39ab7780836a050f5a38c35\n\u003e Reviewed-on: https://webrtc-review.googlesource.com/c/108580\n\u003e Reviewed-by: Oskar Sundbom \u003cossu@webrtc.org\u003e\n\u003e Reviewed-by: Danil Chapovalov \u003cdanilchap@webrtc.org\u003e\n\u003e Commit-Queue: Ilya Nikolaevskiy \u003cilnik@webrtc.org\u003e\n\u003e Cr-Commit-Position: refs/heads/master@{#25430}\n\nTBR\u003ddanilchap@webrtc.org,ilnik@webrtc.org,ossu@webrtc.org\n\nChange-Id: I208a659379b1075258ee94613e42afd9aebe4754\nNo-Presubmit: true\nNo-Tree-Checks: true\nNo-Try: true\nBug: webrtc:9905\nReviewed-on: https://webrtc-review.googlesource.com/c/108623\nReviewed-by: Ilya Nikolaevskiy \u003cilnik@webrtc.org\u003e\nCommit-Queue: Ilya Nikolaevskiy \u003cilnik@webrtc.org\u003e\nCr-Commit-Position: refs/heads/master@{#25435}",
  "tree_diff": [
    {
      "type": "modify",
      "old_id": "e3d34727f6b96fe71d73cc4f4df894aecdcee6b3",
      "old_mode": 33188,
      "old_path": "audio/channel_send.cc",
      "new_id": "cc8b1f78428aed2929a9e6dcca84922eef470cad",
      "new_mode": 33188,
      "new_path": "audio/channel_send.cc"
    },
    {
      "type": "modify",
      "old_id": "b2aad1e7cb989c29864e89e9f90dbb3a014bca76",
      "old_mode": 33188,
      "old_path": "modules/rtp_rtcp/source/rtcp_sender.cc",
      "new_id": "67fdf34d6f20804e8de74e83b183d009a65f33a0",
      "new_mode": 33188,
      "new_path": "modules/rtp_rtcp/source/rtcp_sender.cc"
    }
  ]
}