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gerrit-public.fairphone.software / platform / external / webrtc / ad62792c5dc49c896df56ab1319b5b8556af875a / call / BUILD.gn
  1. ad62792 Fixing hidden dependencies. by Mirko Bonadei · 7 years ago
  2. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  3. c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
  4. 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
  5. fd6c091 Delete deprecated constructor of SendSideCongestionController. by Niels Möller · 7 years ago
  6. f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  7. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  8. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/BUILD.gn]
  9. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  10. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  11. 334f9e6 Tracking mock_paced_sender.h with a GN target by mbonadei · 7 years ago
  12. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  13. 95c8f65 Now that https://codereview.webrtc.org/3003643002 is landed we can by mbonadei · 7 years ago
  14. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  15. f3f5c0e Change ThreadChecker to SequencedTaskChecker in internal::Call by eladalon · 7 years ago
  16. b332917 Rename RsidResolutionObserver to SsrcBindingObserver. by Steve Anton · 7 years ago
  17. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  18. 5166e54 Tracking mock_process_thread with a GN target by mbonadei · 7 years ago
  19. e2173d9 Only one implementation of MockRtpPacketSink once by eladalon · 7 years ago
  20. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  21. c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
  22. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  23. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  24. a52722f Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 7 years ago
  25. 0e7e786 Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 7 years ago
  26. cb83bdf Create RtcpDemuxer. Capabilities: by eladalon · 7 years ago
  27. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  28. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  29. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 7 years ago
  30. 760a076 Create unit tests for RtpDemuxer by eladalon · 7 years ago
  31. c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
  32. eed52bf New class RtxReceiveStream. by nisse · 7 years ago
  33. e4bcd6d New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 7 years ago
  34. 2d9d21f Add untracked headers in modules/rtp_rtcp by danilchap · 8 years ago
  35. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
  36. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  37. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  38. 81c79f5 Creating webrtc:video_stream_api by mbonadei · 8 years ago
  39. e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
  40. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  41. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  42. 37e99fd Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  43. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  44. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  45. 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 · 8 years ago
  46. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  47. 16ccfdf Reland Move fake_audio_device to its own target. by perkj · 8 years ago
  48. 2f1a555 Enable GN check for webrtc/call by kjellander · 8 years ago
  49. 38cc1d6 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  50. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  51. 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
  52. 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
  53. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  54. 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
  55. 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
  56. 0b5a26a Refactor webrtc_perf_tests into several source_sets. by ehmaldonado · 8 years ago
  57. 7250b39 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  58. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  59. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  60. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  61. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  62. bf6a45b Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency by charujain · 8 years ago
  63. 76648da Add FlexfecReceiveStream. by brandtr · 8 years ago
  64. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  65. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  66. 89a3a1a Moved Gn target rtc_event_log to one directory above. by charujain · 8 years ago
  67. b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  68. e9cac75 Reenabled the RtcEventLog unittests by skvlad · 8 years ago
  69. e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 8 years ago
  70. 7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 8 years ago
  71. 38a2132 GN: Introduce templates. by ehmaldonado · 8 years ago
  72. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  73. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  74. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  75. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  76. 0208322 GN: Add video_engine_tests by Peter Boström · 8 years ago
  77. 14897d0 Add missing dependencies on audio, video and call to the new GN files. by katrielc · 8 years ago
  78. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  79. 0e7e259 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 9 years ago
  80. 0c478b3 Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  81. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago