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gerrit-public.fairphone.software / platform / external / webrtc / 4bac9c53da9988741d59753c2d789adb94de5e68 / talk / media / base / fakemediaengine.h
  1. 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
  2. eefbc3b Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ ) by torbjorng · 9 years ago
  3. 1c0bb38 - Remove AudioTrackRenderer. by solenberg · 9 years ago
  4. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  5. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  6. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
  7. 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 9 years ago
  8. 4a3ccad Remove SetAudioDelayOffset() and friends. by solenberg · 9 years ago
  9. 61e933e Remove ChannelManager::GetCapabilities() by solenberg · 9 years ago
  10. 7d17336 Remove the [Un]RegisterVoiceProcessor() API. by Fredrik Solenberg · 9 years ago
  11. c1a1b35 Remove the SetLocalMonitor() API. by solenberg · 9 years ago
  12. 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 9 years ago
  13. b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 9 years ago
  14. 709ed67 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels. by Fredrik Solenberg · 9 years ago
  15. 1dd98f3 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) by solenberg · 9 years ago
  16. 66f4339 Remove [Voice|Video]MediaChannel::GetOptions(). by solenberg · 9 years ago
  17. bb741b3 Remove GetOutputScaling from VoiceMediaChannel. by solenberg · 9 years ago
  18. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  19. c232096 Remove cricket::VideoProcessor and AddVideoProcessor() functionality by Magnus Jedvert · 9 years ago
  20. c28a896 VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation by Jelena Marusic · 9 years ago
  21. 9a416bd Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2 by Fredrik Solenberg · 9 years ago
  22. ccb49e7 Remove Soundclip handling from libjingle. by Fredrik Solenberg · 9 years ago
  23. 4b60c73 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE. by Fredrik Solenberg · 9 years ago
  24. 9478437 rtc::Buffer improvements by Karl Wiberg · 10 years ago
  25. 56d5028 Remove SignalCaptureStateChange from MediaEngine. by Peter Thatcher · 10 years ago
  26. 77f0e3f Remove GetStartCaptureFormat and some related code. by Peter Thatcher · 10 years ago
  27. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
  28. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  29. 058b1f1 Remove GetReceiveBandwidthEstimatorStats. by pbos@webrtc.org · 10 years ago
  30. f1f0d9a Remove WebRtcVideoEngine::SetVoiceEngine. by pbos@webrtc.org · 10 years ago
  31. 586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
  32. 40b276e Cleanup little things found when refactoring. by pthatcher@webrtc.org · 10 years ago
  33. 19b4741 Removing unused method GetDefaultVideoEncoderConfig. by andresp@webrtc.org · 10 years ago
  34. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  35. 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
  36. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  37. 1ecbe45 (Auto)update libjingle 77689511-> 77696841 by buildbot@webrtc.org · 10 years ago
  38. 97abeee (Auto)update libjingle 77263371-> 77296420 by buildbot@webrtc.org · 10 years ago
  39. 992febb (Auto)update libjingle 74873066-> 74873164 by buildbot@webrtc.org · 10 years ago
  40. 818b7b3 (Auto)update libjingle 74825084-> 74825992 by buildbot@webrtc.org · 10 years ago
  41. 3740d74 (Auto)update libjingle 73927658-> 73927775 by buildbot@webrtc.org · 10 years ago
  42. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  43. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  44. 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 11 years ago
  45. a7b9818 Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). by henrike@webrtc.org · 11 years ago
  46. ef22151 Revert 5590 "description" by xians@webrtc.org · 11 years ago
  47. 2643805 description by henrike@webrtc.org · 11 years ago
  48. b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
  49. a8910d2 Update talk to 60094938. by wu@webrtc.org · 11 years ago
  50. 4b26e2e Update libjingle to 59676287 by sergeyu@chromium.org · 11 years ago
  51. a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  52. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  53. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  54. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  55. a27be8e Update libjingle to CL 53398036. by mallinath@webrtc.org · 11 years ago
  56. cadf904 Update talk to 51664136. by wu@webrtc.org · 11 years ago
  57. d64719d Update libjingle to 50191337. by wu@webrtc.org · 11 years ago
  58. 1e09a71 Update talk folder to revision=49952949 by henrike@webrtc.org · 11 years ago
  59. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago