)]}'
{
  "commit": "b9386f3f09ba0f38320f6a9e65dfe2992f433cb5",
  "tree": "1d953f15dfed2d70a85d14d1dc2f1c62b1fcb5c3",
  "parents": [
    "df2dce4a189ab64ebf020f166723b01f1cf3c423"
  ],
  "author": {
    "name": "Philipp Hancke",
    "email": "philipp.hancke@googlemail.com",
    "time": "Thu May 14 10:24:38 2026"
  },
  "committer": {
    "name": "webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com",
    "email": "webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com",
    "time": "Thu May 14 11:43:30 2026"
  },
  "message": "srtp: add UseCryptex API to SrtpSession and SrtpTransport\n\nAdds the UseCryptex(enable, require, sending_session) API on SrtpSession\nand UseCryptex(enable, require) on SrtpTransport. Wires libsrtp\u0027s\nsrtp_set_stream_use_cryptex through DoSetKey, and adds a require-cryptex\npost-decrypt check via ParseRtpExtensionProfile (new helper in\nmodules/rtp_rtcp/source/rtp_util) for the cases libsrtp does not yet\nenforce; see https://github.com/cisco/libsrtp/pull/805.\n\nNo callers yet; the cryptex SDP negotiation wires this up in a follow-up.\n\nBug: webrtc:455813732\nChange-Id: I407400d42912fb303c93c715e7110dafcc3d0cab\nReviewed-on: https://webrtc-review.googlesource.com/c/src/+/470600\nReviewed-by: Harald Alvestrand \u003chta@webrtc.org\u003e\nReviewed-by: Tomas Gunnarsson \u003ctommi@webrtc.org\u003e\nCommit-Queue: Philipp Hancke \u003cphilipp.hancke@googlemail.com\u003e\nCr-Commit-Position: refs/heads/main@{#47707}\n",
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