Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / pc / BUILD.gn
  1. 1d03a75 Remove cricket::RtpTransceiverDirection by Steve Anton · 7 years ago
  2. 3163867 Reland "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
  3. a4ecf55 Revert "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
  4. 6c7ec32 SetRemoteDescriptionObserverInterface added. by Henrik Boström · 7 years ago
  5. 872cf38 Fix some dependencies for peerconnection_and_implicit_call_api. by Edward Lemur · 7 years ago
  6. f2d7beb Created the DtlsSrtpTransport. by Zhi Huang · 7 years ago
  7. 6e634bf Add RtpTransceiverInterface and implementing class by Steve Anton · 7 years ago
  8. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  9. 32df86e Remove deprecated CreatePeerConnectionFactory() overloads by Karl Wiberg · 7 years ago
  10. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  11. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  12. c4faa9c Remove QUIC transport/data channel by Steve Anton · 7 years ago
  13. 8a63f78 Rewrite the remaining few WebRtcSession tests. by Steve Anton · 7 years ago
  14. da6c095 Rewrite WebRtcSession data channel tests as PeerConnection tests by Steve Anton · 7 years ago
  15. 6f25b09 Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Steve Anton · 7 years ago
  16. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  17. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  18. b49b661 Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  19. 096e367 Rewrite WebRtcSession BUNDLE tests as PeerConnection tests by Steve Anton · 7 years ago
  20. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  21. f1c6db1 Rewrite WebRtcSession ICE tests as PeerConnection tests by Steve Anton · 7 years ago
  22. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  23. c5bb00b PeerConnection end-to-end test with a non-builtin codec by Karl Wiberg · 7 years ago
  24. 933d8b0 Reland "Added PeerConnectionObserver::OnRemoveTrack." by Henrik Boström · 7 years ago
  25. 6c0c55c Revert "Added PeerConnectionObserver::OnRemoveTrack." by Alex Loiko · 7 years ago
  26. ba97ba7 Added PeerConnectionObserver::OnRemoveTrack. by Henrik Boström · 7 years ago
  27. 6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
  28. 97a9f76 Add sdputils.h with useful functions for working with session descriptions by Steve Anton · 7 years ago
  29. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  30. 94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 7 years ago
  31. 58b0316 Expose new video codec factories in the PeerConnectionFactory API by Magnus Jedvert · 7 years ago
  32. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  33. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/BUILD.gn]
  34. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  35. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  36. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  37. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  38. c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
  39. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  40. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  41. 4dde3df Move SrtpSession and tests to their own files. by zstein · 7 years ago
  42. 4583db4 Enable -Wunused-function warning everywhere. by Henrik Kjellander · 7 years ago
  43. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  44. ab97e18 Fix the binary size regression on Chromium Windows. by zhihuang · 7 years ago
  45. 130ca7e Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) by zhihuang · 7 years ago
  46. c2e208a Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) by zhihuang · 7 years ago
  47. 9ed1609 Try to fix the binary size increase issue on Chromium. by zhihuang · 7 years ago
  48. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  49. 2b3aa14 Fix Chromium style checker warnings for MockAudioDecoder by kwiberg · 7 years ago
  50. 7d9a55b enabling `gn check` on the whole WebRTC repo by mbonadei · 7 years ago
  51. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  52. 9087d49 Enabling 'gn check' on webrtc/video. by mbonadei · 7 years ago
  53. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  54. 1e060c6 Enabling 'gn check' on webrtc/sdk by mbonadei · 7 years ago
  55. d60d06a Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) by ilnik · 7 years ago
  56. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 7 years ago
  57. 716d7ac Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) by guidou · 7 years ago
  58. c42f540 Move video_encoder.h and video_decoder.h to /api and create GN targets for them by ilnik · 7 years ago
  59. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 7 years ago
  60. eaa9c1d Remove HAVE_SRTP define and unmaintained code. by jbauch · 7 years ago
  61. dfcab72 Reland: Improve testing of SRTP external auth code paths. by jbauch · 7 years ago
  62. d81f121 Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ ) by jbauch · 7 years ago
  63. ac170d5 Improve testing of SRTP external auth code paths. by jbauch · 7 years ago
  64. d48f488 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 7 years ago
  65. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  66. 804c1af Move trackmediainfomap files from api/ to pc/. by deadbeef · 7 years ago
  67. 9aa3f0a Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 7 years ago
  68. 69dc7db Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 7 years ago
  69. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 7 years ago
  70. da25006 Fixed public_deps for libjingle_peerconnection{,_api} by ossu · 8 years ago
  71. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  72. 40610e2 Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. by deadbeef · 8 years ago
  73. 0d8ade5 Remove remnants of libsrtp1 by mattdr · 8 years ago
  74. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  75. 7ba3051 Delete unused class cricket::MediaSinkInterface, and mediasink.h. by nisse · 8 years ago
  76. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  77. 705ecc5 GN: Change group deps to public_deps. by kjellander · 8 years ago
  78. e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 8 years ago
  79. 7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 8 years ago
  80. 38a2132 GN: Introduce templates. by ehmaldonado · 8 years ago
  81. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  82. 142f8c5 GN: Add rtc_pc_unittests by kjellander · 8 years ago
  83. c76dc95 Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 8 years ago
  84. 4d167e5 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #5 id:80001 of https://codereview.webrtc.org/1979933002/ ) by kjellander · 8 years ago
  85. 164e978 Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 8 years ago
  86. fb1dd43 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:20001 of https://codereview.webrtc.org/1973313002/ ) by kjellander · 8 years ago
  87. c8d848b Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 8 years ago
  88. 8744cf6 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:140001 of https://codereview.webrtc.org/1929633002/ ) by kjellander · 8 years ago
  89. 4d02a35 GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 8 years ago