)]}'
{
  "commit": "bb095aa99bb42ccd3d42ff392ca2c04154982a51",
  "tree": "e8958e29f3d91dabb687afc110d0fd93a3c0f92f",
  "parents": [
    "689b5874d4fb5ee1767aef0d05e0a929ac2ea46e"
  ],
  "author": {
    "name": "Niels Möller",
    "email": "nisse@webrtc.org",
    "time": "Thu Aug 30 13:46:50 2018"
  },
  "committer": {
    "name": "Commit Bot",
    "email": "commit-bot@chromium.org",
    "time": "Mon Sep 03 07:28:39 2018"
  },
  "message": "Allow send bitrate \u003c start bitrate in RampUpTest.\n\nPrimarily, this is intended to reduce flakyness of\nRampUpTest.AudioTransportSequenceNumber. We shouldn\u0027t expect audio\nsend rate \u003e\u003d 300 kbps at all time in these tests. And in general, if\nit\u0027s at all relevant to test that bitrate doesn\u0027t drop below the start\nbitrate, a perf test isn\u0027t the right place for that.\n\nA run of\n\n./third_party/gtest-parallel/gtest-parallel  -r 1000 -w 1000 \\\n   --gtest_filter\u003dRampUpTest.AudioTransportSequenceNumber \\\n   out/Release/webrtc_perf_tests\n\npasses when I ran it locally after this change, but fails around 4 out\nof 1000 times before the change.\n\nBug: webrtc:8878\nChange-Id: I08614ce5683c9ba6fe4b72bfde83e6a81445a59b\nReviewed-on: https://webrtc-review.googlesource.com/96900\nReviewed-by: Sebastian Jansson \u003csrte@webrtc.org\u003e\nCommit-Queue: Niels Moller \u003cnisse@webrtc.org\u003e\nCr-Commit-Position: refs/heads/master@{#24523}",
  "tree_diff": [
    {
      "type": "modify",
      "old_id": "1b2177258bcab6c146c204fa3e89d1a6e75120a4",
      "old_mode": 33188,
      "old_path": "call/rampup_tests.cc",
      "new_id": "17b729629a7a0857d9318e1f833b8348a3affdda",
      "new_mode": 33188,
      "new_path": "call/rampup_tests.cc"
    }
  ]
}