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gerrit-public.fairphone.software / platform / external / webrtc / 2ae140ae7e34bd1e102ca832a14b7b0adbb72d15 / pc / mediasession.cc
  1. 45cc890 Assorted logging pedantry by Jonas Olsson · 7 years ago
  2. 8e545ee Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32." by Tommi · 7 years ago
  3. 6780c51 Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32. by Joachim Bauch · 7 years ago
  4. e831b8c Add MSID signaling compatibility for Unified Plan endpoints by Steve Anton · 7 years ago
  5. 3518e7b Add the rejected TransportInfo when creating an answer. by Zhi Huang · 7 years ago
  6. ad7bffc Parameterize PeerConnection media tests for Unified Plan by Steve Anton · 7 years ago
  7. fa2260d Add support for data channels with Unified Plan by Steve Anton · 7 years ago
  8. dcc3c02 Add support for JSEP offer/answer with transceivers by Steve Anton · 7 years ago
  9. b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
  10. 5adfafd Make ContentInfo/ContentDescription slightly more ergonomic by Steve Anton · 7 years ago
  11. 5634427 Remove unused properties from MediaContentDescription by Steve Anton · 7 years ago
  12. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  13. 1d03a75 Remove cricket::RtpTransceiverDirection by Steve Anton · 7 years ago
  14. 6f36747 Use local codec parameters in the answer. by Zhi Huang · 7 years ago
  15. 7aee3d5 Fix ortc_api circular deps. by Patrik Höglund · 7 years ago
  16. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  17. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  18. 80cfb52 RTC_CHECK'ing content type before static_casting descriptions. by Taylor Brandstetter · 7 years ago
  19. 1c34974 Fixing invalid calls to FindMatchingCodec. by Taylor Brandstetter · 7 years ago
  20. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  21. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  22. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/mediasession.cc]
  23. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  24. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 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. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  29. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  30. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  31. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  32. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  33. 8b7e9ad Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings. by deadbeef · 7 years ago
  34. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  35. 2f425aa Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 7 years ago
  36. eaa9c1d Remove HAVE_SRTP define and unmaintained code. by jbauch · 7 years ago
  37. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  38. b789253 Accept SDP with TRANSPORT attributes missing from bundled m= sections. by deadbeef · 8 years ago
  39. 21e4e0b Delete webrtc/base/common.h by nisse · 8 years ago
  40. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 8 years ago
  41. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  42. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  43. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  44. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  45. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  46. 352444f RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  47. 03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  48. f823ede Negotiate H264 profiles in SDP by magjed · 8 years ago
  49. b05fa24 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  50. 3cf8ece Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  51. 9f71ec5 Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  52. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  53. 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 8 years ago
  54. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 8 years ago
  55. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  56. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  57. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 8 years ago
  58. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  59. dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 8 years ago
  60. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 8 years ago
  61. cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 8 years ago
  62. 8c011e5 Simple lint fixes by terelius · 8 years ago
  63. d713e86 Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ ) by zhihuang · 8 years ago
  64. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 8 years ago
  65. b7f425a Accept all the media profiles required by JSEP. by zhihuang · 8 years ago
  66. 5f0b83b Enabling rtcp-rsize negotiation and fixing some issues with it. by Taylor Brandstetter · 8 years ago
  67. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 8 years ago
  68. 6ec641b Fixing some issues with payload type mappings. by Taylor Brandstetter · 8 years ago
  69. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  70. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  71. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  72. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/mediasession.cc]
  73. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  74. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  75. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
  76. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  77. b5cb19b Fixing direction attribute in answer for non-RTP protocols. by deadbeef · 9 years ago
  78. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  79. 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
  80. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  81. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  82. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  83. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  84. 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
  85. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  86. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  87. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  88. 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
  89. 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
  90. a5b273a Fixing problems with RTP extension ID conflict resolution by deadbeef · 9 years ago
  91. 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
  92. 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
  93. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  94. 083b73f Use std::string references instead of copying contents. by jbauch · 9 years ago
  95. f393829 Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used. by deadbeef · 9 years ago
  96. 2e7a098 Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc. by Noah Richards · 9 years ago
  97. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 9 years ago
  98. a747093 After another round of reviews. by lally@webrtc.org · 10 years ago
  99. ec97c65 Attempt on read-only acceptance of -12. by lally@webrtc.org · 10 years ago
  100. 586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago