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gerrit-public.fairphone.software / platform / external / webrtc / f6515cd0e3a11d3893242cbc8313b0fc902aedcc / pc / mediasession.cc
  1. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  2. 80cfb52 RTC_CHECK'ing content type before static_casting descriptions. by Taylor Brandstetter · 7 years ago
  3. 1c34974 Fixing invalid calls to FindMatchingCodec. by Taylor Brandstetter · 7 years ago
  4. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/mediasession.cc]
  7. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  8. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  9. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  10. 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
  11. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  12. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  13. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  14. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  15. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  16. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  17. 8b7e9ad Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings. by deadbeef · 7 years ago
  18. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  19. 2f425aa Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 8 years ago
  20. eaa9c1d Remove HAVE_SRTP define and unmaintained code. by jbauch · 8 years ago
  21. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  22. b789253 Accept SDP with TRANSPORT attributes missing from bundled m= sections. by deadbeef · 8 years ago
  23. 21e4e0b Delete webrtc/base/common.h by nisse · 8 years ago
  24. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 8 years ago
  25. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  26. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  27. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  28. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  29. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  30. 352444f RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  31. 03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  32. f823ede Negotiate H264 profiles in SDP by magjed · 8 years ago
  33. b05fa24 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  34. 3cf8ece Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  35. 9f71ec5 Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  36. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  37. 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 8 years ago
  38. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 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. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  43. dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 8 years ago
  44. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 8 years ago
  45. cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 8 years ago
  46. 8c011e5 Simple lint fixes by terelius · 8 years ago
  47. d713e86 Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ ) by zhihuang · 9 years ago
  48. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  49. b7f425a Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  50. 5f0b83b Enabling rtcp-rsize negotiation and fixing some issues with it. by Taylor Brandstetter · 9 years ago
  51. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  52. 6ec641b Fixing some issues with payload type mappings. by Taylor Brandstetter · 9 years ago
  53. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  54. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  55. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  56. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/mediasession.cc]
  57. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  58. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  59. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
  60. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  61. b5cb19b Fixing direction attribute in answer for non-RTP protocols. by deadbeef · 9 years ago
  62. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  63. 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
  64. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  65. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  66. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  67. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  68. 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
  69. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  70. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  71. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  72. 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
  73. 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
  74. a5b273a Fixing problems with RTP extension ID conflict resolution by deadbeef · 9 years ago
  75. 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
  76. 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
  77. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  78. 083b73f Use std::string references instead of copying contents. by jbauch · 9 years ago
  79. f393829 Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used. by deadbeef · 9 years ago
  80. 2e7a098 Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc. by Noah Richards · 9 years ago
  81. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 10 years ago
  82. a747093 After another round of reviews. by lally@webrtc.org · 10 years ago
  83. ec97c65 Attempt on read-only acceptance of -12. by lally@webrtc.org · 10 years ago
  84. 586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
  85. 5ad4178 Move the Jingle-specific network code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  86. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  87. f15dee6 Check if a datachannel in the current local description is an sctp channel before assuming rtp. by tommi@webrtc.org · 10 years ago
  88. 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
  89. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  90. 742922b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 10 years ago
  91. 7d4891d Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  92. c172320 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. by jiayl@webrtc.org · 10 years ago
  93. 52055a2 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  94. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  95. 56d8e05 A followup to r6828 to fix a condition check in mediasession.cc. by jiayl@webrtc.org · 10 years ago
  96. e7d47a1 Maintain the order of the m-lines in CreateOffer and CreateAnswer. by jiayl@webrtc.org · 10 years ago
  97. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  98. ff1b1bf When creating an answer, takes the codec preference from the offer. by wu@webrtc.org · 10 years ago
  99. 8dcd43c Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 10 years ago
  100. 9c16c39 Sets the SCTP port codec in the native SessionDescription. by jiayl@webrtc.org · 10 years ago