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gerrit-public.fairphone.software / platform / external / webrtc / ede9ca5a24c2b83f2bce182153b9569fa8608648 / pc / peerconnection_integrationtest.cc
  1. ede9ca5 Rewrite WebRtcSession ICE integration tests as PeerConnection tests by Steve Anton · 7 years ago
  2. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  3. bdcee28 TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  4. 604427b Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort" by Guido Urdaneta · 7 years ago
  5. b23ed7f TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  6. 8c0f7a7 Add GetRemoteAudioSSLCertificate() to PeerConnection by Steve Anton · 7 years ago
  7. 94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 7 years ago
  8. 4e2deab Only return stats for the most recent unsignaled audio stream. by deadbeef · 7 years ago
  9. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  10. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  11. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/peerconnection_integrationtest.cc]
  12. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
  13. b1a15d7 In PC integration tests, create tracks/streams with random IDs. by deadbeef · 7 years ago
  14. 4389b4d Add a PeerConnection integration test for adding an audio track mid-call by deadbeef · 7 years ago
  15. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  16. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  17. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  18. 8b7e9ad Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings. by deadbeef · 7 years ago
  19. f816493 Add media related stats (audio level etc.) to unsignaled streams. by zhihuang · 7 years ago
  20. 98e186c Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 7 years ago
  21. 7145280 Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver. by deadbeef · 7 years ago
  22. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  23. 30952b4 Add "ice-option:trickle" to generated offers/answers. by deadbeef · 7 years ago
  24. d8ad788 Adding integration test for unsignaled inbound RTP stream stats. by deadbeef · 8 years ago
  25. 2f425aa Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 8 years ago
  26. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  27. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  28. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  29. c964d0b Fixing some case-sensitive codec name comparisons. by deadbeef · 8 years ago
  30. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 8 years ago