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webrtc / src.git / f3ee3b7478713a723e886d7a5aa3d4d41a0ca4bf / pc / webrtcsession_unittest.cc
  1. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  2. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  3. b49b661 Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  4. 096e367 Rewrite WebRtcSession BUNDLE tests as PeerConnection tests by Steve Anton · 7 years ago
  5. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  6. ede9ca5 Rewrite WebRtcSession ICE integration tests as PeerConnection tests by Steve Anton · 7 years ago
  7. f1c6db1 Rewrite WebRtcSession ICE tests as PeerConnection tests by Steve Anton · 7 years ago
  8. 6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
  9. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 8 years ago
  10. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  11. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/webrtcsession_unittest.cc]
  12. 2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 8 years ago
  13. 6d64e9a Remove JsepSessionDescription's string Initialize method by Steve Anton · 8 years ago
  14. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 8 years ago
  15. 1cc5fc3e Fix places that trigger no-unused-lambda-capture by eladalon · 8 years ago
  16. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 8 years ago
  17. 3c74766 Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ ) by olka · 8 years ago
  18. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 8 years ago
  19. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  20. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  21. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  22. 98e186c Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 8 years ago
  23. 7eaa4ea Delete method MessageQueue::set_socketserver by nisse · 8 years ago
  24. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  25. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  26. 81bf7b0 Pass ownership of candidate to PeerConnection::OnIceCandidate by jbauch · 8 years ago
  27. 38989e5 Parse the connection data in SDP (c= line). by zhihuang · 8 years ago
  28. 6dfd53a Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange by zstein · 8 years ago
  29. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  30. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  31. 94a2f21 Increase STUN RTOs to work better on poor networks, such as 2G networks. by pthatcher · 8 years ago
  32. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  33. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  34. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/webrtcsession_unittest.cc]
  35. f33491e Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ ) by deadbeef · 8 years ago
  36. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  37. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  38. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  39. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  40. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  41. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  42. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  43. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  44. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  45. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  46. df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
  47. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  48. 277b250 Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
  49. 4dfb8ce Make the default value of rtcp-mux policy to required. by zhihuang · 8 years ago
  50. 669d69b Use rtc::PacketTransportInterface in WebrtcSession unit test. by johan · 8 years ago
  51. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  52. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  53. b60a819 Fixing inconsistency with behavior of `ClearGettingPorts`. by deadbeef · 9 years ago
  54. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 9 years ago
  55. 907abe4 Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of https://codereview.webrtc.org/2166873002/ ) by deadbeef · 9 years ago
  56. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 9 years ago
  57. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 9 years ago
  58. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 9 years ago
  59. 3d70fef Remove DtlsIdentityStoreInterface, it is no longer used. by hbos · 9 years ago
  60. 5622c5e If continual gathering is enabled, by Honghai Zhang · 9 years ago
  61. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  62. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  63. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  64. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  65. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  66. 14f97f5 Adding IceConfig option to assume TURN/TURN candidate pairs will work. by deadbeef · 9 years ago
  67. 13d5db3 Revert of Adding IceConfig option to assume TURN/TURN candidate pairs will work. (patchset #9 id:160001 of https://codereview.webrtc.org/2063823008/ ) by honghaiz · 9 years ago
  68. 8e6134e Adding IceConfig option to assume TURN/TURN candidate pairs will work. by Taylor Brandstetter · 9 years ago
  69. 14461d4 Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent by deadbeef · 9 years ago
  70. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
  71. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
  72. f5f03e8 Reland of: Improving the fake clock and using it to fix a flaky STUN timeout test. by deadbeef · 9 years ago
  73. f83a94a Revert of Improving the fake clock and using it to fix a flaky STUN timeout test. (patchset #10 id:180001 of https://codereview.webrtc.org/2024813004/ ) by deadbeef · 9 years ago
  74. ffbe0e1 Improving the fake clock and using it to fix a flaky STUN timeout test. by deadbeef · 9 years ago
  75. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  76. d79599d Turning FakeDtlsIdentityStore into FakeRTCCertificateGenerator. by Henrik Boström · 9 years ago
  77. d03c23b Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 9 years ago
  78. d7973cc Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ ) by hbos · 9 years ago
  79. 400781a Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 9 years ago
  80. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
  81. e9021a3 Propogate network-worker thread split to api by danilchap · 9 years ago
  82. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  83. a1c3035 Relanding: Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 9 years ago
  84. c55fb30 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ ) by deadbeef · 9 years ago
  85. 48e9d05 Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 9 years ago
  86. 3fe372d Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 9 years ago
  87. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  88. 3a33465 Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer. by zhihuang · 9 years ago
  89. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago
  90. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  91. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  92. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  93. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  94. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  95. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  96. 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 9 years ago
  97. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  98. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  99. 7fb69db Reland the CL to remove candidates when doing continual gathering by Honghai Zhang · 9 years ago
  100. 6f59a4f Revert of Remove candidates when doing continual gathering (patchset #15 id:560001 of https://codereview.webrtc.org/1648813004/ ) by tommi · 9 years ago