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gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / pc / webrtcsdp_unittest.cc
  1. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  2. aba85d1 Resolve circular dependency in rtc_media_base. by Patrik Höglund · 7 years ago
  3. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  4. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 7 years ago
  5. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  6. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  7. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/webrtcsdp_unittest.cc]
  8. 3e8016e Ignore "b=AS:-1" instead of treating as a hard error. by deadbeef · 7 years ago
  9. bc88c6b Reject negative values for "b=AS". by deadbeef · 7 years ago
  10. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  11. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  12. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  13. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  14. f184138 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 7 years ago
  15. 121cabb Fix webrtcsdp_unittest. by ehmaldonado · 7 years ago
  16. 38989e5 Parse the connection data in SDP (c= line). by zhihuang · 7 years ago
  17. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 7 years ago
  18. a4549d6 Fix SDP parsing crash due to missing track ID in "a=msid". by deadbeef · 7 years ago
  19. 90f1e1e Fixing SDP parsing crash due to invalid port numbers. by deadbeef · 7 years ago
  20. aa4b077 Simplify IsFmtpParam according to RFC 4855. by ossu · 7 years ago
  21. e1405ad Removed double-special-casing of ISAC in libjingle and WebRtcVoE. by ossu · 8 years ago
  22. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/webrtcsdp_unittest.cc]
  23. 7bcdb69 Ignore ufrag/password in "a=candidate" lines in SDP. by deadbeef · 8 years ago
  24. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  25. 12771a1 Relax parsing of a=bundle-only with a nonzero port. by deadbeef · 8 years ago
  26. b236257 Fixing integer overflow when parsing bandwidth attribute. by deadbeef · 8 years ago
  27. 25ed435 Implement parsing/serialization of a=bundle-only. by deadbeef · 8 years ago
  28. 509e4fe Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) by magjed · 8 years ago
  29. eacbaea Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) by magjed · 8 years ago
  30. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  31. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  32. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  33. 7e146cb Fixing heap read overflow when "sctp-port" is in a video description. by deadbeef · 8 years ago
  34. 2d8d23e RFC 3984 sprop-parameter-sets SDP unit test by johan · 8 years ago
  35. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  36. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  37. 62a216e Don't write spaces after semicolons in FMTP lines. by hta · 8 years ago
  38. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 8 years ago
  39. a6b9944 Generate FMTP parameters for the H.264 codec. by hta · 8 years ago
  40. a0c44ea Add 16-bit network id to the candidate signaling. by honghaiz · 8 years ago
  41. 5de6b75 If MSID is encoded in both ways, make the SSRC-level one take priority. by Taylor Brandstetter · 8 years ago
  42. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 8 years ago
  43. 9d3584c Implementing unified plan encoding of msid. by deadbeef · 8 years ago
  44. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 8 years ago
  45. b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
  46. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (99%) from talk/app/webrtc/webrtcsdp_unittest.cc]
  47. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  48. 46eed76 Removing "candidates" attribute from TransportDescription. by deadbeef · 8 years ago
  49. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 9 years ago
  50. 3f7219b Fixing issue where description contains empty ICE ufrag/pwd. by deadbeef · 9 years ago
  51. a54a080 Add ufrag to the ICE candidate signaling. by honghaiz · 9 years ago
  52. bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 9 years ago
  53. a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
  54. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  55. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  56. 69f5760 Added parsing of either space or colon for sctp-port. by lally · 9 years ago
  57. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  58. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  59. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  60. 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
  61. 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
  62. 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
  63. 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
  64. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  65. 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
  66. c0c3a86 Prevent JS from bypassing RTP data channel bandwidth limitation. by Peter Thatcher · 9 years ago
  67. 144d018 fix indent on tokenize_first function signatures by Donald Curtis · 9 years ago
  68. 0e07f92 Split fmtp on semicolons not spaces as per RFC6871 by Donald Curtis · 9 years ago
  69. 3480728 Swap decl-terms from juberti@ review. by lally@webrtc.org · 9 years ago
  70. 3630085 Tested equiv classes of DTLS/SCTP. by lally@webrtc.org · 9 years ago
  71. 91d5230 Renamed string and test. by lally@webrtc.org · 9 years ago
  72. c7848b7 Added a separate DTLS/SCTP test. by lally@webrtc.org · 9 years ago
  73. d7b6165 Made DTLS/SCTP equivalent to UDP/DTLS/SCTP when comparing session descs in tests. by lally@webrtc.org · 9 years ago
  74. ec97c65 Attempt on read-only acceptance of -12. by lally@webrtc.org · 9 years ago
  75. d324546 Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : by pkasting@chromium.org · 9 years ago
  76. a744a28 Templatize and clean up codec wildcards. by jlmiller@webrtc.org · 9 years ago
  77. f9b5c1b Removing CELT. by minyue@webrtc.org · 9 years ago
  78. 57ac2c8 Default destination used by c line should be IPv4 only to avoid parsing error in legacy client. by guoweis@webrtc.org · 9 years ago
  79. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  80. 61c1247 Fix a case where empty candidate id is used by guoweis@webrtc.org · 10 years ago
  81. 8af1104 Avoid reading past end of string in GetLine. by decurtis@webrtc.org · 10 years ago
  82. 950c518 Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  83. 55360ae Revert "Add adapter_type into Candidate object." by guoweis@webrtc.org · 10 years ago
  84. aaf02cc Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  85. fb108b5 Revert r7885. by pbos@webrtc.org · 10 years ago
  86. 8c9d79a Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
  87. d105cc8 Change dummy address to use 0.0.0.0 instead of :: by perkj@webrtc.org · 10 years ago
  88. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  89. c9d6d14 patch from issue 25469004 by pthatcher@webrtc.org · 10 years ago
  90. 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
  91. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  92. ddb85ab Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  93. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  94. e999bd0 Removing ASSERT for tcp candidate for port 0 and 9, as Android clients by mallinath@webrtc.org · 10 years ago
  95. 7ec3f9f Fix a bug in parsing IceCandidate with IPV6 address. by jiayl@webrtc.org · 10 years ago
  96. 2d60c5e Encoding and Decoding of TCP candidates as defined in RFC 6544. by mallinath@webrtc.org · 10 years ago
  97. e7d47a1 Maintain the order of the m-lines in CreateOffer and CreateAnswer. by jiayl@webrtc.org · 10 years ago
  98. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  99. 4c3e991 Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be: by wu@webrtc.org · 10 years ago
  100. ec9f5fb Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE]. by wu@webrtc.org · 10 years ago