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gerrit-public.fairphone.software / platform / external / webrtc / 6753795409a242d2cac9288e2d62256cd111f07b / pc / peerconnectioninterface_unittest.cc
  1. 6753795 Built in video codec factories. by Anders Carlsson · 6 years ago
  2. c79268f Add IsClosed checks to various PeerConnection methods by Steve Anton · 6 years ago
  3. a2d6067 Reland "Add thread checker to PortAllocator and its subclasses and fix a bug causing memory contention by threads." by Qingsi Wang · 6 years ago
  4. 3dc4106 Revert "Add thread checker to PortAllocator and its subclasses and fix a bug" by Patrik Höglund · 6 years ago
  5. fc43d11 Add thread checker to PortAllocator and its subclasses and fix a bug by Qingsi Wang · 6 years ago
  6. 83d676b Bug fix for applying a remote description twice without stream IDs. by Seth Hampson · 6 years ago
  7. fd350d7 By default, don't use SRTP_AES128_CM_SHA1_32 protection profile. by Taylor Brandstetter · 6 years ago
  8. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 6 years ago
  9. 644fde4 Add nullptr check in SctpTransport. by Zhi Huang · 6 years ago
  10. 5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 6 years ago
  11. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 6 years ago
  12. 191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 6 years ago
  13. f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 6 years ago
  14. bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 6 years ago
  15. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 6 years ago
  16. 13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 6 years ago
  17. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 6 years ago
  18. 5a26a3a Remove public sync_label from StreamParams by Steve Anton · 6 years ago
  19. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  20. 6e22137 Enable Unified Plan tests that were blocked on the stats collector by Steve Anton · 7 years ago
  21. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  22. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  23. 36da6ff Parameterize PeerConnection interface tests for Unified Plan by Steve Anton · 7 years ago
  24. 57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
  25. ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
  26. 56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
  27. 8e545ee Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32." by Tommi · 7 years ago
  28. 6780c51 Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32. by Joachim Bauch · 7 years ago
  29. 93a8439 Bind the structured ICE logging with P2PTransportChannel. by Qingsi Wang · 7 years ago
  30. 1d7ecd2 Rename a few MediaConfig::Video flags for consistency. by Niels Möller · 7 years ago
  31. 6539f69 Add VideoSendStream::Config::EncoderSettings::experiment_cpu_load_estimator. by Niels Möller · 7 years ago
  32. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  33. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  34. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  35. 62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
  36. 2d6c76a Switch to using AddTrack with stream labels by Steve Anton · 7 years ago
  37. 8906187 Pivot generation of stats to iterate senders/receivers by Harald Alvestrand · 7 years ago
  38. b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
  39. 74cefe1 Removing dependency on JsepTransport from DtlsTransport tests. by Taylor Brandstetter · 7 years ago
  40. a3a92c2 Replace string type with SdpType enum by Steve Anton · 7 years ago
  41. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  42. b95fd2c Optional: Use nullopt and implicit construction in /pc/peerconnectioninterface_unittest.cc by Oskar Sundbom · 7 years ago
  43. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  44. de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
  45. b2d355e Reland: Reject the description with fewer m= sections. by Zhi Huang · 7 years ago
  46. 6f25b09 Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Steve Anton · 7 years ago
  47. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  48. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  49. b49b661 Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  50. 096e367 Rewrite WebRtcSession BUNDLE tests as PeerConnection tests by Steve Anton · 7 years ago
  51. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  52. 1b0eae3 Don't call deprecated CreatePeerConnectionFactory() overloads by Karl Wiberg · 7 years ago
  53. 919dc2e Removes fallback from Linux PulseAudio to ALSA. by henrika · 7 years ago
  54. 589ae45 Revert "Reject the subsequent offer with fewer m= sections." by Tommi · 7 years ago
  55. a8264db Reject the subsequent offer with fewer m= sections. by Zhi Huang · 7 years ago
  56. f1c6db1 Rewrite WebRtcSession ICE tests as PeerConnection tests by Steve Anton · 7 years ago
  57. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  58. 9e6565b Fix PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed by Elad Alon · 7 years ago
  59. 94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 7 years ago
  60. 02e7a19 Remove unnecessary video factory references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  61. 835cc0c Remove unnecessary audio references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  62. b19012e Remove the support of fallback from DTLS to SDES. by zhihuang · 7 years ago
  63. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  64. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  65. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/peerconnectioninterface_unittest.cc]
  66. 2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 7 years ago
  67. db45ca8 Change PeerConnection test helpers to take unique_ptr by Steve Anton · 7 years ago
  68. 141aacb Fix the Chromium crash when creating an answer without a remote description. by zhihuang · 7 years ago
  69. d7850b2 Use fake audio device in peerconnectioninterface_unittest.cc. by deadbeef · 7 years ago
  70. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  71. 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
  72. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  73. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  74. d21eab3 Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 7 years ago
  75. ec390b5 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 7 years ago
  76. 038834f Reinstate "Add additional check when setting RTCConfiguration" by Steve Anton · 7 years ago
  77. e725159 Reland of Make the default ctor of rtc::Thread, protected by tommi · 7 years ago
  78. 26d5e2e Revert "Add additional check when setting RTCConfiguration" by Magnus Jedvert · 7 years ago
  79. 8110bed Add additional check when setting RTCConfiguration by Steve Anton · 7 years ago
  80. a117b04 Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ ) by charujain · 7 years ago
  81. a8a3515 Make the default ctor of rtc::Thread, protected. by tommi · 7 years ago
  82. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 7 years ago
  83. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 7 years ago
  84. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  85. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  86. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  87. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  88. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  89. 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 · 7 years ago
  90. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 7 years ago
  91. 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 · 7 years ago
  92. 98e186c Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 7 years ago
  93. 9a6f4d4 Get tests working on systems that only support IPv6. by deadbeef · 7 years ago
  94. 528b793 Update comments for removal of MediaController. by nisse · 7 years ago
  95. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  96. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  97. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  98. 30952b4 Add "ice-option:trickle" to generated offers/answers. by deadbeef · 7 years ago
  99. a1a040a Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 7 years ago
  100. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 7 years ago