Roll src/third_party/webrtc 519d74a5fcd8..cc189177a63a (15 commits)

git log 519d74a5fcd8..cc189177a63a --date=short --no-merges --format='%ad %ae %s'
2019-05-20 Revert "Improve spec compliance of SetStreamIDs in RtpSenderInterface"
2019-05-20 Remove unused dependency.
2019-05-20 Roll chromium_revision 243a2094e7..f5d370078e (660868:660984)
2019-05-20 Update android tests to use single argument PeerConnectionFactory factory
2019-05-20 Delete configuration of unused transport_sequence_number_allocator
2019-05-20 Fix test names in
2019-05-20 Reland "Delete deprecated rtc::Thread default constructor"
2019-05-18 Introduce peer connection level webrtc video quality tests.
2019-05-17 Add CreateDatagram to MediaTransportFactory
2019-05-17 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video.
2019-05-17 Remove non-source sources from binary targets
2019-05-17 Add juberti@ to webrtc root owners
2019-05-17 Roll chromium_revision 4c9872694a..243a2094e7 (660753:660868)
2019-05-17 Adds feedback generator.
2019-05-17 Improve spec compliance of SetStreamIDs in RtpSenderInterface

Created with:
  gclient setdep -r src/third_party/webrtc@cc189177a63a

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Change-Id: Iaec69a158a6b7d9c8ef4cd9a589bc16bb71f3a93
Reviewed-by: chromium-autoroll <>
Commit-Queue: chromium-autoroll <>
Cr-Commit-Position: refs/heads/master@{#661368}
1 file changed