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gerrit-public.fairphone.software / platform / external / webrtc / f3850f6933c468625511fc9e2df9aa4b9fefed89 / call / BUILD.gn
  1. f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  2. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  3. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/BUILD.gn]
  4. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  5. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  6. 334f9e6 Tracking mock_paced_sender.h with a GN target by mbonadei · 7 years ago
  7. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  8. 95c8f65 Now that https://codereview.webrtc.org/3003643002 is landed we can by mbonadei · 7 years ago
  9. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  10. f3f5c0e Change ThreadChecker to SequencedTaskChecker in internal::Call by eladalon · 7 years ago
  11. b332917 Rename RsidResolutionObserver to SsrcBindingObserver. by Steve Anton · 7 years ago
  12. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  13. 5166e54 Tracking mock_process_thread with a GN target by mbonadei · 7 years ago
  14. e2173d9 Only one implementation of MockRtpPacketSink once by eladalon · 7 years ago
  15. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  16. c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
  17. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  18. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  19. a52722f Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 7 years ago
  20. 0e7e786 Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 7 years ago
  21. cb83bdf Create RtcpDemuxer. Capabilities: by eladalon · 7 years ago
  22. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  23. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  24. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 7 years ago
  25. 760a076 Create unit tests for RtpDemuxer by eladalon · 7 years ago
  26. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
  27. eed52bf New class RtxReceiveStream. by nisse · 7 years ago
  28. e4bcd6d New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 7 years ago
  29. 2d9d21f Add untracked headers in modules/rtp_rtcp by danilchap · 7 years ago
  30. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 7 years ago
  31. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  32. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  33. 81c79f5 Creating webrtc:video_stream_api by mbonadei · 7 years ago
  34. e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 7 years ago
  35. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago
  36. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  37. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 7 years ago
  38. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  39. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  40. c5d62e2 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 7 years ago
  41. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
  42. 16ccfdf Reland Move fake_audio_device to its own target. by perkj · 8 years ago
  43. 2f1a555 Enable GN check for webrtc/call by kjellander · 8 years ago
  44. 38cc1d6 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  45. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  46. 9aa3f0a Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 8 years ago
  47. 69dc7db Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 8 years ago
  48. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  49. 021eef3 Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) by ehmaldonado · 8 years ago
  50. 5fbcd22 Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) by danilchap · 8 years ago
  51. 0b5a26a Refactor webrtc_perf_tests into several source_sets. by ehmaldonado · 8 years ago
  52. 7250b39 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  53. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  54. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  55. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  56. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  57. bf6a45b Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency by charujain · 8 years ago
  58. 76648da Add FlexfecReceiveStream. by brandtr · 8 years ago
  59. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  60. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  61. 89a3a1a Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  62. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  63. e9cac75 Reenabled the RtcEventLog unittests by skvlad · 8 years ago
  64. e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 8 years ago
  65. 7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 8 years ago
  66. 38a2132 GN: Introduce templates. by ehmaldonado · 8 years ago
  67. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  68. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  69. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  70. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  71. 0208322 GN: Add video_engine_tests by Peter Boström · 8 years ago
  72. 14897d0 Add missing dependencies on audio, video and call to the new GN files. by katrielc · 8 years ago
  73. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  74. 0e7e259 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 9 years ago
  75. 0c478b3 Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  76. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago