0583b286e4
RTCIceCandidatePairStats.transport_id is set to the related RTCTransportStats' id. Unittest for RTCIceCandidatePairStats is updated to do EXPECT_EQ between actual and an expected hardcoded dictionary. The previous way of testing, ExpectReportContainsCandidatePair, is removed. (ExpectReportContainsCandidate still exist, we might want to replace this by EXPECT_EQ testing in a follow up.) Unittest for RTCTransportStats is similarly updated and ExpectReportContainsTransportStats is removed. A bug was uncovered where the "rtcp_connection_info.best_connection = true" case was not tested (a copy of rtcp_connection_info was used in the test, modifying that had no affect on the test) - fixed. rtcstats_integrationtest.cc updated to take transport_id into account. In order to reuse an updated version of expected_rt[c]p_transport in the unittest, timestamps are ignored by RTCStats::operator==. BUG=chromium:627816, chromium:653873, chromium:653873, webrtc:6755 Review-Url: https://codereview.webrtc.org/2527113002 Cr-Commit-Position: refs/heads/master@{#15316}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.