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gerrit-public.fairphone.software / platform / external / webrtc / c6ce9c59385102e766913dcfc5eb69cb5cf135ba / api / mediastreaminterface.h
  1. c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 6 years ago
  2. 13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 6 years ago
  3. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  4. 2a5ce2b Fix clang style errors in rtp_rtcp and dependant targets by Danil Chapovalov · 7 years ago
  5. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  6. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  7. 21eb9fc Make the old GetStats interface on AudioProcessorInterface impure. by Ivo Creusen · 7 years ago
  8. 3a23374 Reland "Remove the aec_quality_min metric." by Gustaf Ullberg · 7 years ago
  9. a3fad93 Revert "Remove the aec_quality_min metric." by Mirko Bonadei · 7 years ago
  10. 99b1bd1 Remove the aec_quality_min metric. by Gustaf Ullberg · 7 years ago
  11. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  12. ae02609 Add parallel stats interface with optional stats to APM. by Ivo Creusen · 7 years ago
  13. e2d6a06 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  14. 1af3d82 Revert "Reland "Clean up libjingle API dependencies."" by Henrik Kjellander · 7 years ago
  15. 9185aca Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  16. 581df61 Revert "Reland "Clean up libjingle API dependencies."" by Patrik Höglund · 7 years ago
  17. 5117b04 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  18. 7bcfc3b Revert "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  19. 57fb315 Clean up libjingle API dependencies. by Patrik Höglund · 7 years ago
  20. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  21. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/mediastreaminterface.h]
  22. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  23. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  24. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  25. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 7 years ago
  26. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 7 years ago
  27. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  28. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  29. f93752a Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 7 years ago
  30. 61b22dd Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ ) by nisse · 7 years ago
  31. 3870a07 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ ) by nisse · 7 years ago
  32. 6e6a485 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ ) by nisse · 7 years ago
  33. d71ebd7 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ ) by nisse · 7 years ago
  34. aec49d2 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) by nisse · 7 years ago
  35. 713a3bb Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 7 years ago
  36. 8d60a94 Replace NULL with nullptr or null in webrtc/api/. by deadbeef · 8 years ago
  37. b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
  38. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  39. af91689 Move VideoFrame and related declarations to webrtc/api/video. by nisse · 8 years ago
  40. 9baddf2 Replace basictypes.h with stddef.h for size_t. by pbos · 8 years ago
  41. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  42. acd935b Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  43. 7341ab8 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ ) by nisse · 8 years ago
  44. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  45. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  46. 859e861 Remove stop method from VideoTrackSourceInterface. by sakal · 8 years ago
  47. a973f95 Remove restart method from VideoTrackSourceInterface. by sakal · 8 years ago
  48. 5d58333 Fix VideoFrame inclusion in mediastreaminterface.h by perkj · 8 years ago
  49. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 8 years ago
  50. 2a8a78c Add AEC filter divergence metric to StatsCollector. by Minyue · 8 years ago
  51. efc3858 Remove deprecated MediaStreamTrackInterface::set_state by perkj · 8 years ago
  52. fcc640f Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 8 years ago
  53. c0d31e9 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 8 years ago
  54. 7ca142e ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy. by perkj · 8 years ago
  55. d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 8 years ago
  56. 8f59762 Delete VideoRendererInterface. by Niels Möller · 8 years ago
  57. c8f952d Propagate MediaStreamSource state to video tracks the same way as audio. by perkj · 8 years ago
  58. f0dcfe2 Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource. by perkj · 8 years ago
  59. 0d3eef2 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 8 years ago
  60. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 8 years ago
  61. db25d2e Make VideoTrack and VideoTrackRenderers implement rtc::VideoSourceInterface. by nisse · 9 years ago
  62. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  63. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/mediastreaminterface.h]
  64. 8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 9 years ago
  65. e73afba New rtc::VideoSinkInterface. by nisse · 9 years ago
  66. 6a062bd Deleted method AudioTrackInterface::GetRenderer. by nisse · 9 years ago
  67. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 9 years ago
  68. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 9 years ago
  69. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  70. 3e1cfa7 Delete unused method webrtc::VideoRendererInterface::SetSize. by nisse · 9 years ago
  71. 6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
  72. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  73. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  74. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  75. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  76. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  77. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  78. c2db810 Remove VideoRendererInterface::CanApplyRotation() by Magnus Jedvert · 9 years ago
  79. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  80. 00c509a Add concept of whether video renderer supports rotation. by guoweis@webrtc.org · 9 years ago
  81. f9a75d9 Revert "Add concept of whether video renderer supports rotation." by guoweis@webrtc.org · 9 years ago
  82. 0ad4893 Add concept of whether video renderer supports rotation. by guoweis@webrtc.org · 9 years ago
  83. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  84. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  85. b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 10 years ago
  86. 40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 11 years ago
  87. b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
  88. 0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 11 years ago
  89. e749c9e Update libjingle to 61514460 by xians@webrtc.org · 11 years ago
  90. 67ee6b9 Update talk to 60923971 by mallinath@webrtc.org · 11 years ago
  91. 967bfff Update talk to 52534915. by wu@webrtc.org · 11 years ago
  92. 32001ef PeerConnection shutdown-time fixes by fischman@webrtc.org · 11 years ago
  93. 1e09a71 Update talk folder to revision=49952949 by henrike@webrtc.org · 11 years ago
  94. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago