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gerrit-public.fairphone.software / platform / external / webrtc / b65ec395511ed4dc63e99d046cf732ab327695ce / pc / peerconnectioninterface_unittest.cc
  1. 74cefe1 Removing dependency on JsepTransport from DtlsTransport tests. by Taylor Brandstetter · 7 years ago
  2. a3a92c2 Replace string type with SdpType enum by Steve Anton · 7 years ago
  3. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  4. b95fd2c Optional: Use nullopt and implicit construction in /pc/peerconnectioninterface_unittest.cc by Oskar Sundbom · 7 years ago
  5. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  6. de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
  7. b2d355e Reland: Reject the description with fewer m= sections. by Zhi Huang · 7 years ago
  8. 6f25b09 Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Steve Anton · 7 years ago
  9. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  10. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  11. b49b661 Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  12. 096e367 Rewrite WebRtcSession BUNDLE tests as PeerConnection tests by Steve Anton · 7 years ago
  13. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  14. 1b0eae3 Don't call deprecated CreatePeerConnectionFactory() overloads by Karl Wiberg · 7 years ago
  15. 919dc2e Removes fallback from Linux PulseAudio to ALSA. by henrika · 7 years ago
  16. 589ae45 Revert "Reject the subsequent offer with fewer m= sections." by Tommi · 7 years ago
  17. a8264db Reject the subsequent offer with fewer m= sections. by Zhi Huang · 7 years ago
  18. f1c6db1 Rewrite WebRtcSession ICE tests as PeerConnection tests by Steve Anton · 7 years ago
  19. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  20. 9e6565b Fix PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed by Elad Alon · 7 years ago
  21. 94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 7 years ago
  22. 02e7a19 Remove unnecessary video factory references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  23. 835cc0c Remove unnecessary audio references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  24. b19012e Remove the support of fallback from DTLS to SDES. by zhihuang · 7 years ago
  25. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  26. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  27. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/peerconnectioninterface_unittest.cc]
  28. 2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 7 years ago
  29. db45ca8 Change PeerConnection test helpers to take unique_ptr by Steve Anton · 7 years ago
  30. 141aacb Fix the Chromium crash when creating an answer without a remote description. by zhihuang · 7 years ago
  31. d7850b2 Use fake audio device in peerconnectioninterface_unittest.cc. by deadbeef · 7 years ago
  32. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  33. 3c74766 Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ ) by olka · 7 years ago
  34. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  35. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  36. d21eab3 Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 7 years ago
  37. ec390b5 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 7 years ago
  38. 038834f Reinstate "Add additional check when setting RTCConfiguration" by Steve Anton · 8 years ago
  39. e725159 Reland of Make the default ctor of rtc::Thread, protected by tommi · 8 years ago
  40. 26d5e2e Revert "Add additional check when setting RTCConfiguration" by Magnus Jedvert · 8 years ago
  41. 8110bed Add additional check when setting RTCConfiguration by Steve Anton · 8 years ago
  42. a117b04 Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ ) by charujain · 8 years ago
  43. a8a3515 Make the default ctor of rtc::Thread, protected. by tommi · 8 years ago
  44. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 8 years ago
  45. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 8 years ago
  46. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  47. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  48. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  49. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  50. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  51. 4b97980 Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  52. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  53. 9641c13 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  54. 98e186c Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 8 years ago
  55. 9a6f4d4 Get tests working on systems that only support IPv6. by deadbeef · 8 years ago
  56. 528b793 Update comments for removal of MediaController. by nisse · 8 years ago
  57. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  58. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  59. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  60. 30952b4 Add "ice-option:trickle" to generated offers/answers. by deadbeef · 8 years ago
  61. a1a040a Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  62. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 8 years ago
  63. 42a4263 Making candidate pool size behave as decided in JSEP. by deadbeef · 8 years ago
  64. 7f06766 Delete deprecated PeerConnection methods, and corresponding using declarations. by nisse · 8 years ago
  65. 6038e97 Adding RTCErrorOr class to be used by ORTC APIs. by deadbeef · 8 years ago
  66. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  67. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  68. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 8 years ago
  69. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 8 years ago
  70. 63b14b7 Add override declarations to PeerConnectionObserver subclasses, and delete obsolete methods. by nisse · 8 years ago
  71. 1e4e8cb Add CreatePeerConnectionFactory overloads that take audio codec factory args by kwiberg · 8 years ago
  72. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  73. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  74. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/peerconnectioninterface_unittest.cc]
  75. 8662f94 Only set certificate on DTLS transport if fingerprint is found in SDP. by deadbeef · 8 years ago
  76. 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
  77. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  78. 293e926 Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  79. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  80. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  81. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  82. 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
  83. 7a5fa6c Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  84. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  85. 6de92f9 Don't allow changing ICE pool size after SetLocalDescription. by deadbeef · 8 years ago
  86. d1a38b5 Implement the "needs-ice-restart" logic for SetConfiguration. by deadbeef · 8 years ago
  87. 3edec7c Adding error enum to be used by PeerConnectionInterface methods. by deadbeef · 8 years ago
  88. c63b894 Modify the parameter type of PeerConnectionObserver callback OnAddTrack. by zhihuang · 8 years ago
  89. 4dfb8ce Make the default value of rtcp-mux policy to required. by zhihuang · 8 years ago
  90. 81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  91. 46c7389 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  92. 71a1b61 WebRTC: Fix and enable -Woverloaded-virtual warnings. by kjellander · 8 years ago
  93. e9e94c3 Return false if PeerConnection::GetStats() is called on invalid tracks by zhihuang · 8 years ago
  94. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  95. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  96. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  97. bfd398c Add a switch to redetermine role when ICE restarts. by Honghai Zhang · 8 years ago
  98. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  99. 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
  100. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago