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gerrit-public.fairphone.software / platform / external / webrtc / dfe8ca0d430c5d42083b4218baf0ef80e755011f / pc / mediasession_unittest.cc
  1. f8ed561 Remove last mention of speex codec by Henrik Lundin · 7 years ago
  2. fd350d7 By default, don't use SRTP_AES128_CM_SHA1_32 protection profile. by Taylor Brandstetter · 7 years ago
  3. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  4. 5a26a3a Remove public sync_label from StreamParams by Steve Anton · 7 years ago
  5. 9c1fb1e Consider packetization-mode when matching H264 codecs by Steve Anton · 7 years ago
  6. 8e545ee Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32." by Tommi · 7 years ago
  7. 6780c51 Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32. by Joachim Bauch · 7 years ago
  8. 3518e7b Add the rejected TransportInfo when creating an answer. by Zhi Huang · 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. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  16. 98ea2da Removing logging in unit test that was committed accidentally. by Taylor Brandstetter · 7 years ago
  17. 1c34974 Fixing invalid calls to FindMatchingCodec. by Taylor Brandstetter · 7 years ago
  18. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  19. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/mediasession_unittest.cc]
  20. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  21. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  22. 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
  23. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  24. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  25. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  26. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  27. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  28. 8b7e9ad Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings. by deadbeef · 8 years ago
  29. 30952b4 Add "ice-option:trickle" to generated offers/answers. by deadbeef · 8 years ago
  30. eaa9c1d Remove HAVE_SRTP define and unmaintained code. by jbauch · 8 years ago
  31. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 8 years ago
  32. abcef5d Replace std::tr1::tuple by ::testing::tuple. by ehmaldonado · 8 years ago
  33. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  34. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  35. 03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  36. b05fa24 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  37. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 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 · 9 years ago
  41. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
  42. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  43. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  44. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 9 years ago
  45. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  46. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 9 years ago
  47. cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  48. 8c011e5 Simple lint fixes by terelius · 9 years ago
  49. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  50. 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
  51. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  52. b7f425a Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  53. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  54. 6ec641b Fixing some issues with payload type mappings. by Taylor Brandstetter · 9 years ago
  55. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  56. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  57. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/mediasession_unittest.cc]
  58. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  59. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  60. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
  61. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  62. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  63. 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
  64. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  65. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  66. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  67. 3a14bf3 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers. by Henrik Boström · 9 years ago
  68. 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
  69. 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
  70. a5b273a Fixing problems with RTP extension ID conflict resolution by deadbeef · 9 years ago
  71. 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
  72. 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
  73. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  74. 2e7a098 Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc. by Noah Richards · 10 years ago
  75. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 10 years ago
  76. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  77. f15dee6 Check if a datachannel in the current local description is an sctp channel before assuming rtp. by tommi@webrtc.org · 10 years ago
  78. 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
  79. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  80. 742922b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 10 years ago
  81. 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
  82. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  83. e7d47a1 Maintain the order of the m-lines in CreateOffer and CreateAnswer. by jiayl@webrtc.org · 10 years ago
  84. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  85. ff1b1bf When creating an answer, takes the codec preference from the offer. by wu@webrtc.org · 11 years ago
  86. 8dcd43c Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 11 years ago
  87. 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 11 years ago
  88. b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 11 years ago
  89. 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 11 years ago
  90. 32f485b Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  91. cecfd18 Update talk to 55821645. by wu@webrtc.org · 11 years ago
  92. 97077a3 Update libjingle to 55618622. Update libyuv to r826. by wu@webrtc.org · 11 years ago
  93. 19f27e6 Update talk to 54527154. by mallinath@webrtc.org · 11 years ago
  94. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  95. 0be6aa0 Update talk to 51314459 by sergeyu@chromium.org · 11 years ago
  96. 28654cb Update talk folder to revision=49713299. by henrike@webrtc.org · 11 years ago
  97. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 12 years ago