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gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / pc / peerconnection.cc
  1. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  2. 1d03a75 Remove cricket::RtpTransceiverDirection by Steve Anton · 7 years ago
  3. 9158ef6 Reland "Add AddTransceiver and GetTransceivers to PeerConnection" by Steve Anton · 7 years ago
  4. 8b13f96 Revert "Add AddTransceiver and GetTransceivers to PeerConnection" by Steve Anton · 7 years ago
  5. f93d280 Add AddTransceiver and GetTransceivers to PeerConnection by Steve Anton · 7 years ago
  6. 3163867 Reland "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
  7. a4ecf55 Revert "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
  8. 6c7ec32 SetRemoteDescriptionObserverInterface added. by Henrik Boström · 7 years ago
  9. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  10. 9b28a03 Optional: Use nullopt and implicit construction in /pc/peerconnection.cc by Oskar Sundbom · 7 years ago
  11. 9e6fd2b Add streams() to RtpReceiverInterface and implementations. by Henrik Boström · 7 years ago
  12. de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
  13. 79e7960 Add SDP semantics option to RTCConfiguration by Steve Anton · 7 years ago
  14. 4171afb Use RtpTransceivers in PeerConnection by Steve Anton · 7 years ago
  15. 33c5c7f Revert "Encode log events periodically instead of for every event." by Zhi Huang · 7 years ago
  16. b154c27 Encode log events periodically instead of for every event. by Bjorn Terelius · 7 years ago
  17. fdb9201 Avoid use-after-move in PeerConnection::Set[Local/Remote]Description. by Henrik Boström · 7 years ago
  18. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  19. d25da37 Inline various trivial methods from the WebRtcSession merge by Steve Anton · 7 years ago
  20. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  21. ba81867 Prepare WebRtcSession to be merged into PeerConnection by Steve Anton · 7 years ago
  22. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  23. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  24. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  25. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  26. d5585ca Move almost all references from WebRtcSession to PeerConnection by Steve Anton · 7 years ago
  27. c4faa9c Remove QUIC transport/data channel by Steve Anton · 7 years ago
  28. da6c095 Rewrite WebRtcSession data channel tests as PeerConnection tests by Steve Anton · 7 years ago
  29. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  30. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  31. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  32. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  33. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  34. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  35. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  36. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  37. d6b4819 PeerConnection::StartRtcEventLog: Improve callback memory safety by Karl Wiberg · 7 years ago
  38. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  39. bdcee28 TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  40. 933d8b0 Reland "Added PeerConnectionObserver::OnRemoveTrack." by Henrik Boström · 7 years ago
  41. 6c0c55c Revert "Added PeerConnectionObserver::OnRemoveTrack." by Alex Loiko · 7 years ago
  42. ba97ba7 Added PeerConnectionObserver::OnRemoveTrack. by Henrik Boström · 7 years ago
  43. 604427b Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort" by Guido Urdaneta · 7 years ago
  44. b23ed7f TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  45. 83ccca1 Create and use RtcEventLogOutput for output by Elad Alon · 7 years ago
  46. 8c0f7a7 Add GetRemoteAudioSSLCertificate() to PeerConnection by Steve Anton · 7 years ago
  47. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  48. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
  49. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  50. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  51. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/peerconnection.cc]
  52. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  53. 248fd4f Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread ( https://codereview.webrtc.org/3007473002/ ) by eladalon · 7 years ago
  54. 23814b7 Revert of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #4 id:200001 of https://codereview.webrtc.org/3005153002/ ) by eladalon · 7 years ago
  55. d67cefb Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #1 id:1 of https://codereview.webrtc.org/3010143002/ ) by eladalon · 7 years ago
  56. 141aacb Fix the Chromium crash when creating an answer without a remote description. by zhihuang · 7 years ago
  57. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  58. 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
  59. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  60. d21eab3 Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 7 years ago
  61. ec390b5 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 7 years ago
  62. 038834f Reinstate "Add additional check when setting RTCConfiguration" by Steve Anton · 7 years ago
  63. 300bf8e Reinstate "API for periodically regathering ICE candidates" by Steve Anton · 7 years ago
  64. 3beb207 Revert "API for periodically regathering ICE candidates" by Magnus Jedvert · 7 years ago
  65. 26d5e2e Revert "Add additional check when setting RTCConfiguration" by Magnus Jedvert · 7 years ago
  66. 8110bed Add additional check when setting RTCConfiguration by Steve Anton · 7 years ago
  67. aa41f0c API for periodically regathering ICE candidates by Steve Anton · 7 years ago
  68. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  69. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  70. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  71. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  72. 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
  73. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 7 years ago
  74. 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
  75. 3386025 Initialize PeerConnection members in declaration order and destroy them in reverse order. by terelius · 7 years ago
  76. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  77. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 7 years ago
  78. 81bf7b0 Pass ownership of candidate to PeerConnection::OnIceCandidate by jbauch · 7 years ago
  79. 42a4263 Making candidate pool size behave as decided in JSEP. by deadbeef · 7 years ago
  80. 7f06766 Delete deprecated PeerConnection methods, and corresponding using declarations. by nisse · 7 years ago
  81. b09b3f9 Add the option to disable IPv6 ICE candidates on WiFi. by zhihuang · 7 years ago
  82. 6dfd53a Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange by zstein · 7 years ago
  83. c1b57a1 Test field trial group with startswith rather than equals. by sprang · 7 years ago
  84. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  85. 6038e97 Adding RTCErrorOr class to be used by ORTC APIs. by deadbeef · 7 years ago
  86. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 7 years ago
  87. 9dd77ba Clarifying error messages in ParseIceServerUrl for invalid transport parameters. by zstein · 7 years ago
  88. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 7 years ago
  89. 5107246 Allow applications to limit the ICE check rate through RTCConfiguration by skvlad · 7 years ago
  90. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 7 years ago
  91. 7ce109a Replace the easy cases of VERIFY usage. by nisse · 7 years ago
  92. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/peerconnection.cc]
  93. e8abe3e Revert of New method StatsObserver::OnCompleteReports, passing ownership. (patchset #2 id:20001 of https://codereview.webrtc.org/2584553002/ ) by nisse · 8 years ago
  94. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  95. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  96. 293e926 Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  97. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  98. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  99. 0483362 Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  100. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago