)]}'
{
  "commit": "db208317eb8c9275653fb7168484f5d25e34e374",
  "tree": "fcd8fc01bb968a3a129baa0acd086f252cb8d356",
  "parents": [
    "9d21eb3e3eb279d587f9726a7a03e168b688e340"
  ],
  "author": {
    "name": "Jakob Ivarsson",
    "email": "jakobi@webrtc.org",
    "time": "Fri Jan 27 14:13:22 2023"
  },
  "committer": {
    "name": "WebRTC LUCI CQ",
    "email": "webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com",
    "time": "Fri Jan 27 15:46:32 2023"
  },
  "message": "Update RTP timestamp based on capture timestamp when audio send stream is resumed.\n\nThis removes the previous approach where we continued to update the timestamp when the capturer is running but the send stream is stopped in favor of a more general approach that also works when the capturer is paused.\n\nSome assumptions for this change to be correct: input sample rate and frame size will be the same before/after the stream is paused.\n\nBug: webrtc:12397\nChange-Id: I3b03741cd6d3285cbc9aee3893800729852e6cfa\nReviewed-on: https://webrtc-review.googlesource.com/c/src/+/291526\nCommit-Queue: Jakob Ivarsson‎ \u003cjakobi@webrtc.org\u003e\nReviewed-by: Henrik Lundin \u003chenrik.lundin@webrtc.org\u003e\nCr-Commit-Position: refs/heads/main@{#39213}\n",
  "tree_diff": [
    {
      "type": "modify",
      "old_id": "5b0858a76a149353cda74176a6f6d7f707258bc8",
      "old_mode": 33188,
      "old_path": "audio/channel_send.cc",
      "new_id": "26b5bf452daff4e682090173102a25fcb97fac24",
      "new_mode": 33188,
      "new_path": "audio/channel_send.cc"
    },
    {
      "type": "modify",
      "old_id": "50d8368d4a21b63298ab1947715776f2d4536ef7",
      "old_mode": 33188,
      "old_path": "audio/channel_send_unittest.cc",
      "new_id": "97882b93b7f82baf1ee4de840ded08b38afbec8f",
      "new_mode": 33188,
      "new_path": "audio/channel_send_unittest.cc"
    }
  ]
}