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webrtc / src / webrtc / 68337ca2dc9568763d00a933cba190c0a3bb2dab / call / BUILD.gn
  1. 9ae9fd9 Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 8 years ago
  2. ca3693c Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 8 years ago
  3. 5af9557 Create RtcpDemuxer. Capabilities: by eladalon · 8 years ago
  4. 04ef6b0 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  5. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  6. 2d1de82 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  7. baa253b Create unit tests for RtpDemuxer by eladalon · 8 years ago
  8. 30fa632 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
  9. 002b533 New class RtxReceiveStream. by nisse · 8 years ago
  10. 78bbf3a New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 8 years ago
  11. cd2f080 Add untracked headers in modules/rtp_rtcp by danilchap · 8 years ago
  12. 29a1f8c This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
  13. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  14. fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  15. 279bdc9 Creating webrtc:video_stream_api by mbonadei · 8 years ago
  16. 9cdb538 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
  17. cddf701 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  18. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  19. af8f6e5 Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  20. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  21. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  22. 7950bab 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
  23. 22de8f3 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  24. b880884 Reland Move fake_audio_device to its own target. by perkj · 8 years ago
  25. 388c1df Enable GN check for webrtc/call by kjellander · 8 years ago
  26. 6d5d814 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  27. 9551ae0 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  28. 90cc2a9 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
  29. df5ec64 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
  30. 5c59d8f Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  31. 5445b07 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
  32. a0ac1d2 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
  33. 2d606cb Refactor webrtc_perf_tests into several source_sets. by ehmaldonado · 8 years ago
  34. 02a6431 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  35. 31212ba Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  36. 017ebe5 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  37. d85b51d Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  38. 7713f33 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  39. 5daa11f Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency by charujain · 8 years ago
  40. 4e97439 Add FlexfecReceiveStream. by brandtr · 8 years ago
  41. a6c6623 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  42. e59b6ff Moved RtcEventLog files from call/ to logging/ by skvlad · 9 years ago
  43. a9eeb97 Moved Gn target rtc_event_log to one directory above. by charujain · 9 years ago
  44. 2cf29b5 GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 9 years ago
  45. 0db32c3 Reenabled the RtcEventLog unittests by skvlad · 9 years ago
  46. 80e6692 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 9 years ago
  47. 1d49219 GN Templates: Move common_config to the template. by ehmaldonado · 9 years ago
  48. f0532f3 GN: Introduce templates. by ehmaldonado · 9 years ago
  49. 3f65eaf This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  50. cf82062 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  51. bdd04aa Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  52. cd76a16 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  53. c9da01c GN: Add video_engine_tests by Peter Boström · 9 years ago
  54. eaeea5b Add missing dependencies on audio, video and call to the new GN files. by katrielc · 9 years ago
  55. ead3cf2 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  56. 1b78cc3 Move BitrateAllocator from BitrateController logic to Call. by mflodman · 9 years ago
  57. b62f41a Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  58. bf9f73c Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago