Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / ae810c10b4eeda122b65c7ef3e7af8878b69ede0 / pc / mediasession.h
  1. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  2. 365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
  3. 1b8773d Negotiate the MID header extension for Unified Plan by Steve Anton · 6 years ago
  4. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 6 years ago
  5. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 6 years ago
  6. e831b8c Add MSID signaling compatibility for Unified Plan endpoints by Steve Anton · 7 years ago
  7. ad7bffc Parameterize PeerConnection media tests for Unified Plan by Steve Anton · 7 years ago
  8. fa2260d Add support for data channels with Unified Plan by Steve Anton · 7 years ago
  9. afd8e8c Move MediaContentDescription into sessiondescription.h by Steve Anton · 7 years ago
  10. 4ab68ee Move sessiondescription.h/cc from p2p/base to pc/ by Steve Anton · 7 years ago
  11. 6e2e7ce Reland "Move JsepTransport from p2p/base to pc/." by Taylor Brandstetter · 7 years ago
  12. 8424acd Revert "Move JsepTransport from p2p/base to pc/." by Oleh Prypin · 7 years ago
  13. 4770fd9 Move JsepTransport from p2p/base to pc/. by Taylor Brandstetter · 7 years ago
  14. 5634427 Remove unused properties from MediaContentDescription by Steve Anton · 7 years ago
  15. f72ab83 Remove transceiver direction getter/setter by Steve Anton · 7 years ago
  16. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  17. 73da79c Step 1 to remove MediaContentDirection by Steve Anton · 7 years ago
  18. 1d03a75 Remove cricket::RtpTransceiverDirection by Steve Anton · 7 years ago
  19. 1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
  20. 7aee3d5 Fix ortc_api circular deps. by Patrik Höglund · 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/mediasession.h]
  24. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  25. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  26. 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
  27. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  28. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  29. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  30. 38989e5 Parse the connection data in SDP (c= line). by zhihuang · 8 years ago
  31. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  32. b789253 Accept SDP with TRANSPORT attributes missing from bundled m= sections. by deadbeef · 8 years ago
  33. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 8 years ago
  34. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  35. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  36. 57fd726 Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ ) by deadbeef · 8 years ago
  37. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  38. 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 8 years ago
  39. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  40. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  41. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 8 years ago
  42. a1c548b Add RtpHeaderExtension to avoid client breakage by isheriff · 8 years ago
  43. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  44. dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 8 years ago
  45. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 8 years ago
  46. 8c011e5 Simple lint fixes by terelius · 8 years ago
  47. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 8 years ago
  48. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  49. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  50. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  51. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  52. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed from talk/session/media/mediasession.h]
  53. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  54. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  55. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  56. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  57. 318166b Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 9 years ago
  58. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  59. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  60. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  61. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 9 years ago
  62. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  63. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  64. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  65. 2159b89 Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by Peter Thatcher · 9 years ago
  66. 5bdafd4 Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."" by minyuel · 9 years ago
  67. 081f34b Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots." by Peter Thatcher · 9 years ago
  68. fa30180 Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by pthatcher · 9 years ago
  69. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  70. a747093 After another round of reviews. by lally@webrtc.org · 10 years ago
  71. ec97c65 Attempt on read-only acceptance of -12. by lally@webrtc.org · 10 years ago
  72. 586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
  73. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  74. 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  75. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  76. 742922b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 10 years ago
  77. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  78. e7d47a1 Maintain the order of the m-lines in CreateOffer and CreateAnswer. by jiayl@webrtc.org · 10 years ago
  79. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  80. 8dcd43c Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 10 years ago
  81. 9c16c39 Sets the SCTP port codec in the native SessionDescription. by jiayl@webrtc.org · 10 years ago
  82. b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 11 years ago
  83. 4b26e2e Update libjingle to 59676287 by sergeyu@chromium.org · 11 years ago
  84. cecfd18 Update talk to 55821645. by wu@webrtc.org · 11 years ago
  85. 1112c30 Update libjingle to 53057474. by mallinath@webrtc.org · 11 years ago
  86. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago