Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 6753795409a242d2cac9288e2d62256cd111f07b / pc / peerconnectioninterface_unittest.cc
  1. 42a4263 Making candidate pool size behave as decided in JSEP. by deadbeef · 8 years ago
  2. 7f06766 Delete deprecated PeerConnection methods, and corresponding using declarations. by nisse · 8 years ago
  3. 6038e97 Adding RTCErrorOr class to be used by ORTC APIs. by deadbeef · 8 years ago
  4. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  5. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  6. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 8 years ago
  7. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 8 years ago
  8. 63b14b7 Add override declarations to PeerConnectionObserver subclasses, and delete obsolete methods. by nisse · 8 years ago
  9. 1e4e8cb Add CreatePeerConnectionFactory overloads that take audio codec factory args by kwiberg · 8 years ago
  10. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  11. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  12. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/peerconnectioninterface_unittest.cc]
  13. 8662f94 Only set certificate on DTLS transport if fingerprint is found in SDP. by deadbeef · 8 years ago
  14. 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
  15. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  16. 293e926 Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  17. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  18. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  19. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  20. 1e23461 Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ ) by deadbeef · 8 years ago
  21. 7a5fa6c Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  22. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  23. 6de92f9 Don't allow changing ICE pool size after SetLocalDescription. by deadbeef · 8 years ago
  24. d1a38b5 Implement the "needs-ice-restart" logic for SetConfiguration. by deadbeef · 8 years ago
  25. 3edec7c Adding error enum to be used by PeerConnectionInterface methods. by deadbeef · 8 years ago
  26. c63b894 Modify the parameter type of PeerConnectionObserver callback OnAddTrack. by zhihuang · 8 years ago
  27. 4dfb8ce Make the default value of rtcp-mux policy to required. by zhihuang · 8 years ago
  28. 81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  29. 46c7389 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  30. 71a1b61 WebRTC: Fix and enable -Woverloaded-virtual warnings. by kjellander · 8 years ago
  31. e9e94c3 Return false if PeerConnection::GetStats() is called on invalid tracks by zhihuang · 8 years ago
  32. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  33. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  34. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  35. bfd398c Add a switch to redetermine role when ICE restarts. by Honghai Zhang · 8 years ago
  36. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  37. 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 · 8 years ago
  38. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  39. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 8 years ago
  40. f8e6577 Add virtual Initialize methods to PortAllocator and NetworkManager. by Taylor Brandstetter · 8 years ago
  41. ba8d433 Revert of Add virtual Initialize methods to PortAllocator and NetworkManager. (patchset #4 id:60001 of https://codereview.webrtc.org/2097653002/ ) by deadbeef · 8 years ago
  42. a6bdb09 Add virtual Initialize methods to PortAllocator and NetworkManager. by Taylor Brandstetter · 8 years ago
  43. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 8 years ago
  44. d79599d Turning FakeDtlsIdentityStore into FakeRTCCertificateGenerator. by Henrik Boström · 8 years ago
  45. d03c23b Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 8 years ago
  46. 6034705 Add a flag to filter out high-cost networks. by honghaiz · 8 years ago
  47. 98cde26 Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel. by Taylor Brandstetter · 8 years ago
  48. d7973cc Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ ) by hbos · 8 years ago
  49. 400781a Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 8 years ago
  50. 417eebe Fixing the behavior of the candidate filter with pooled candidates. by Taylor Brandstetter · 8 years ago
  51. e9021a3 Propogate network-worker thread split to api by danilchap · 8 years ago
  52. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  53. a1c3035 Relanding: Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 8 years ago
  54. c55fb30 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ ) by deadbeef · 8 years ago
  55. 48e9d05 Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 8 years ago
  56. dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 8 years ago
  57. 3fe372d Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 8 years ago
  58. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 8 years ago
  59. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  60. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 8 years ago
  61. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 8 years ago
  62. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  63. 7ff1737 Re-enabling tests that were disabled for Windows debug builds. by Taylor Brandstetter · 8 years ago
  64. 71bdda0 Add RTCConfiguration getter and setter methods. The immediate plan is to move some flags into an embedded MediaConfig (https://codereview.webrtc.org/1818033002/), which will be possible after Chrome is updated to use these new setter methods. by Niels Möller · 8 years ago
  65. d45b95c Making new unit test assertions use the standard timeout. by Taylor Brandstetter · 8 years ago
  66. 85e46a8 Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates by Per · 8 years ago
  67. d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 8 years ago
  68. af510af Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. by nisse · 8 years ago
  69. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  70. 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 8 years ago
  71. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  72. 102362b Truly disable tests. by Stefan Holmer · 8 years ago
  73. 55d6e7c Disable tests due to issue 5659. by Stefan Holmer · 8 years ago
  74. 2bbff99 Helpers in peer connection unit tests: Use scoped_ptr instead of raw pointers by kwiberg · 8 years ago
  75. aac2dea Changed defaults for CreateAnswer in non-constraint mode by hta · 9 years ago
  76. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 9 years ago
  77. a2a49d9 This CL provides interfaces that do not use constraints for by hta · 9 years ago
  78. 0db023a Move suspend_below_min_bitrate from VideoOptions to MediaConfig. by nisse · 9 years ago
  79. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  80. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  81. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  82. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  83. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/peerconnectioninterface_unittest.cc]
  84. dfb769d Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete by perkj · 9 years ago
  85. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  86. 884f585 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  87. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  88. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  89. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  90. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  91. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 9 years ago
  92. 0c7e9f5 Removing webrtc::PortAllocatorFactoryInterface. by Taylor Brandstetter · 9 years ago
  93. bd7d8f7 Adding a MediaStream parameter to createSender. by deadbeef · 9 years ago
  94. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  95. eb45981 Restoring behavior where PeerConnection tracks changes to MediaStreams. by deadbeef · 9 years ago
  96. bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 9 years ago
  97. a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
  98. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  99. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  100. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago