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gerrit-public.fairphone.software / platform / external / webrtc / d42acbb58e1a4c252d221f5205f27efae62c876d / pc / webrtcsdp.cc
  1. b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
  2. 5adfafd Make ContentInfo/ContentDescription slightly more ergonomic by Steve Anton · 7 years ago
  3. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  4. 7aee3d5 Fix ortc_api circular deps. by Patrik Höglund · 7 years ago
  5. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  6. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  7. e2d6a06 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  8. 1af3d82 Revert "Reland "Clean up libjingle API dependencies."" by Henrik Kjellander · 7 years ago
  9. 9185aca Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  10. 581df61 Revert "Reland "Clean up libjingle API dependencies."" by Patrik Höglund · 7 years ago
  11. 5117b04 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  12. 7bcfc3b Revert "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  13. 57fb315 Clean up libjingle API dependencies. by Patrik Höglund · 7 years ago
  14. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 7 years ago
  15. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  16. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/webrtcsdp.cc]
  17. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  18. 3e8016e Ignore "b=AS:-1" instead of treating as a hard error. by deadbeef · 7 years ago
  19. bc88c6b Reject negative values for "b=AS". by deadbeef · 7 years ago
  20. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  21. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  22. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  23. 88af8b4 Fix -Wcomment warning in webrtcsdp.cc by kjellander · 7 years ago
  24. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  25. 38989e5 Parse the connection data in SDP (c= line). by zhihuang · 8 years ago
  26. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  27. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 8 years ago
  28. a4549d6 Fix SDP parsing crash due to missing track ID in "a=msid". by deadbeef · 8 years ago
  29. 90f1e1e Fixing SDP parsing crash due to invalid port numbers. by deadbeef · 8 years ago
  30. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  31. aa4b077 Simplify IsFmtpParam according to RFC 4855. by ossu · 8 years ago
  32. 7d25426 Delete unneeded includes of base/common.h. by nisse · 8 years ago
  33. e1405ad Removed double-special-casing of ISAC in libjingle and WebRtcVoE. by ossu · 8 years ago
  34. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/webrtcsdp.cc]
  35. 7bcdb69 Ignore ufrag/password in "a=candidate" lines in SDP. by deadbeef · 8 years ago
  36. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  37. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  38. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  39. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  40. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  41. 12771a1 Relax parsing of a=bundle-only with a nonzero port. by deadbeef · 8 years ago
  42. b236257 Fixing integer overflow when parsing bandwidth attribute. by deadbeef · 8 years ago
  43. b68cc75 ParseCandidate(): Refactor to fix memcheck false positive. by hnsl · 8 years ago
  44. 277b250 Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
  45. 25ed435 Implement parsing/serialization of a=bundle-only. by deadbeef · 8 years ago
  46. b39db84 Refactoring: Declare cricket::Codec constructors protected. by hta · 8 years ago
  47. b05fa24 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  48. 87d7d77 Add new codec for FlexFEC. by brandtr · 8 years ago
  49. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  50. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  51. 7e146cb Fixing heap read overflow when "sctp-port" is in a video description. by deadbeef · 8 years ago
  52. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 8 years ago
  53. 3e33bfe Fix some sign-compare warnings in webrtc/api. by kjellander · 8 years ago
  54. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  55. 351d77b Update the type and cost of existing networks by Honghai Zhang · 8 years ago
  56. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  57. 62a216e Don't write spaces after semicolons in FMTP lines. by hta · 9 years ago
  58. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  59. a6b9944 Generate FMTP parameters for the H.264 codec. by hta · 9 years ago
  60. a0c44ea Add 16-bit network id to the candidate signaling. by honghaiz · 9 years ago
  61. 7fb69db Reland the CL to remove candidates when doing continual gathering by Honghai Zhang · 9 years ago
  62. 6f59a4f Revert of Remove candidates when doing continual gathering (patchset #15 id:560001 of https://codereview.webrtc.org/1648813004/ ) by tommi · 9 years ago
  63. 84430da When doing candidate re-gathering in the same generation, Remove the existing local candidate on the same network by honghaiz · 9 years ago
  64. 5de6b75 If MSID is encoded in both ways, make the SSRC-level one take priority. by Taylor Brandstetter · 9 years ago
  65. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  66. 9788534 Removing some redundant ostringstreams declarations. by Taylor Brandstetter · 9 years ago
  67. 9d3584c Implementing unified plan encoding of msid. by deadbeef · 9 years ago
  68. e1a0c94 Add network cost as part of the connection ranking. by honghaiz · 9 years ago
  69. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  70. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  71. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (99%) from talk/app/webrtc/webrtcsdp.cc]
  72. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  73. 46eed76 Removing "candidates" attribute from TransportDescription. by deadbeef · 9 years ago
  74. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  75. 3f7219b Fixing issue where description contains empty ICE ufrag/pwd. by deadbeef · 9 years ago
  76. a54a080 Add ufrag to the ICE candidate signaling. by honghaiz · 9 years ago
  77. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  78. 5237aaf Convert usage of ARRAY_SIZE to arraysize. by tfarina · 9 years ago
  79. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  80. 69f5760 Added parsing of either space or colon for sctp-port. by lally · 9 years ago
  81. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  82. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  83. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  84. 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
  85. 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
  86. 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
  87. 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
  88. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  89. a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 9 years ago
  90. 083b73f Use std::string references instead of copying contents. by jbauch · 9 years ago
  91. bb36fdf Remove empty-string comparisons. by pbos · 9 years ago
  92. c0c3a86 Prevent JS from bypassing RTP data channel bandwidth limitation. by Peter Thatcher · 9 years ago
  93. 144d018 fix indent on tokenize_first function signatures by Donald Curtis · 9 years ago
  94. 0e07f92 Split fmtp on semicolons not spaces as per RFC6871 by Donald Curtis · 9 years ago
  95. 019087f Add safeguards against signalling peer-reflexive candidates. by Peter Thatcher · 10 years ago
  96. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 10 years ago
  97. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 10 years ago
  98. a747093 After another round of reviews. by lally@webrtc.org · 10 years ago
  99. 9616196 Merging definitions of IsSctp. by lally@webrtc.org · 10 years ago
  100. 12aa8a6 Post-rebase. by lally@webrtc.org · 10 years ago