)]}'
{
  "commit": "e40a408898b7ea16e59ebc8ea96c6d42c8f8753d",
  "tree": "ff8eb710fdd54219d3b5f2bd224a9de8c36c1a65",
  "parents": [
    "5a076f110ee1c60a62567175b8f231a4489efc23"
  ],
  "author": {
    "name": "Tommi",
    "email": "tommi@webrtc.org",
    "time": "Wed Oct 07 15:29:02 2026"
  },
  "committer": {
    "name": "webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com",
    "email": "webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com",
    "time": "Thu Oct 08 13:18:59 2026"
  },
  "message": "Provide RtpState at construction when restoring an RTP module\n\nWhen a suspended stream was recreated, its previous state was applied\nwith SetRtpState() and SetRtxState() right after construction. This\nrequired state to be mutable that otherwise could be const, which in\nturn required a lock for every read on the encoder queue.\n\n* Add rtp_state and rtx_rtp_state to RtpRtcpInterface::Configuration.\n* Apply it when constructing RTPSender and ModuleRtpRtcpImpl2.\n* RtpVideoSender and AudioSendStream, via ChannelSend, now pass the\n  suspended states at creation.\n* Remove SetStartTimestamp(), SetRtpState() and SetRtxState() from\n  RtpRtcpInterface, ModuleRtpRtcpImpl2, as well as\n  RTPSender::SetRtxRtpState().\n* Deprecate RTPSender::SetTimestampOffset() and SetRtpState(). Once\n  direct users of RTPSender have migrated, they can be removed and the\n  timestamp offset made const.\n\nBug: webrtc:42223727\nChange-Id: Idad26fe1205c1288a3455865dfb79027ed7f83da\nReviewed-on: https://webrtc-review.googlesource.com/c/src/+/507320\nReviewed-by: Danil Chapovalov \u003cdanilchap@webrtc.org\u003e\nCommit-Queue: Tomas Gunnarsson \u003ctommi@webrtc.org\u003e\nCr-Commit-Position: refs/heads/main@{#48831}\n",
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