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gerrit-public.fairphone.software / platform / external / webrtc / fae640003cf028477819309777ecd29a74c25eff / pc / data_channel.cc
  1. 977b265 Reduce some logging at INFO level by moving log statements by Harald Alvestrand · 4 years, 8 months ago
  2. dfbfb46 Return an error when datachannel closes due to network error by Harald Alvestrand · 4 years, 8 months ago
  3. 408cb4b Make SCTPtransport enter "closed" state when DTLStransport does. by Harald Alvestrand · 4 years, 8 months ago
  4. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  5. b689af4 Changes to enable use of DatagramTransport as a data channel transport. by Bjorn A Mellem · 5 years ago
  6. 928e7a3 Make ID of datachannel stats not depend on dc.id by Harald Alvestrand · 5 years ago
  7. f3736ed Datachannel: Use absl::optional for maxRetransmits and maxRetransmitTime. by Harald Alvestrand · 5 years ago
  8. b9bb371 Revert "Don't crash when a datachannel can't get an ID" by Harald Alvestrand · 5 years ago
  9. 77c442c Don't crash when a datachannel can't get an ID by Harald Alvestrand · 5 years ago
  10. e448a3f Update DataChannel bufferedamount implementation. by Marina Ciocea · 5 years ago
  11. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  12. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/datachannel.cc]
  13. 944c755 Use unique_ptr in DataChannel PacketQueue by Steve Anton · 6 years ago
  14. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  15. 175aa2e Implement data channels over media transport. by Bjorn Mellem · 6 years ago
  16. 044a04d Use AsyncInvoker in DataChannel instead of MessageHandler by Steve Anton · 6 years ago
  17. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  18. cdd05f0 Implement proper SCTP data channel closing procedure. by Taylor Brandstetter · 6 years ago
  19. 45cc890 Assorted logging pedantry by Jonas Olsson · 6 years ago
  20. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  21. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  22. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  23. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/datachannel.cc]
  24. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  25. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  26. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  27. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  28. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/datachannel.cc]
  29. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  30. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  31. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  32. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  33. 82ebe02 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
  34. 84ffdee DataChannel[Interface]::[message/bytes]_[sent/received]() added. by hbos · 8 years ago
  35. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 8 years ago
  36. 4cb5b64 Fix for data channels perpetually stuck in "closing" state. by Taylor Brandstetter · 8 years ago
  37. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  38. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 8 years ago
  39. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  40. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 8 years ago
  41. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  42. 3e33bfe Fix some sign-compare warnings in webrtc/api. by kjellander · 8 years ago
  43. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  44. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  45. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  46. 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
  47. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  48. b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
  49. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (98%) from talk/app/webrtc/datachannel.cc]
  50. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  51. ab9b2d1 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ ) by deadbeef · 9 years ago
  52. fc648b6 Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ ) by deadbeef · 9 years ago
  53. 97c3929 Moving MediaStreamSignaling logic into PeerConnection. by deadbeef · 9 years ago
  54. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  55. 0edd50c Support for onbufferedamountlow by bemasc · 9 years ago
  56. 5c6c6e0 Implements TODOs for webrtc::datachannel state management when the SCTP association is congested. Adds missing state variables for each step in the transitions between DataChannelInterface::DataStates (kConnecting, kOpen, etc.), and uses them. by Lally Singh · 9 years ago
  57. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 9 years ago
  58. cceb166 Fix a use-after-free when sending queued messages is aborted for blocked channel. by jiayl@webrtc.org · 10 years ago
  59. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  60. 9b5467e Fix assertion failure when closing data channel, and add a unit test. by bemasc@webrtc.org · 10 years ago
  61. 6ca6190 Fix a SCTP message reordering issue in datachannel.cc. by jiayl@webrtc.org · 10 years ago
  62. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  63. 3edbaaf Ignore empty data in DataChannel::Send to match FF's behavior. by jiayl@webrtc.org · 10 years ago
  64. b43c99d Limits the send and receive buffer by bytes, not by packets. by jiayl@webrtc.org · 10 years ago
  65. 9f8164c Fix two bugs in DataChannel state transition. by jiayl@webrtc.org · 10 years ago
  66. 5dc51fb Closes the DataChannel when the send buffer is full or on transport errors. by jiayl@webrtc.org · 10 years ago
  67. b364016 Revert r6161 "Drop the DataChannel message if it's received when the channel is not open." by jiayl@webrtc.org · 10 years ago
  68. 4f58014 Drop the DataChannel message if it's received when the channel is not open. by jiayl@webrtc.org · 10 years ago
  69. aebb1ad pRevert 5371 "Revert 5367 "Update talk to 59410372."" by henrika@webrtc.org · 11 years ago
  70. 44461fa Revert 5367 "Update talk to 59410372." by henrika@webrtc.org · 11 years ago
  71. 0f3356e Update talk to 59410372. by mallinath@webrtc.org · 11 years ago
  72. a23f0ca Update talk to 56619788 by sergeyu@chromium.org · 11 years ago
  73. 07a6fbe Update talk to 56092586. by wu@webrtc.org · 11 years ago
  74. cecfd18 Update talk to 55821645. by wu@webrtc.org · 11 years ago
  75. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  76. a59696b Update libjingle to 52300956 by sergeyu@chromium.org · 11 years ago
  77. cadf904 Update talk to 51664136. by wu@webrtc.org · 11 years ago
  78. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 11 years ago
  79. 91053e7 Update libjingle to 50654631. by wu@webrtc.org · 11 years ago
  80. a054569 Fix memory leak in datachannel and its test. by wu@webrtc.org · 11 years ago
  81. d64719d Update libjingle to 50191337. by wu@webrtc.org · 11 years ago
  82. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago