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gerrit-public.fairphone.software / platform / external / webrtc / fc47ed6c0524d7ee0bc7947f0ad65fcefda34a29
  1. fc47ed6 rtcp::Rrtr block moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  2. 1aa420b Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead. by asapersson · 9 years ago
  3. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  4. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  5. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  6. a856542 Initial VideoProcessing refactoring. by mflodman · 9 years ago
  7. 2512f44 Roll chromium_revision 292ab9f..4918765 (363376:363393) by kjellander · 9 years ago
  8. c9f1cb8 Roll chromium_revision 72c3265..292ab9f (363365:363376) by kjellander · 9 years ago
  9. 34a7054 Roll chromium_revision 626eecf..72c3265 (363027:363365) by kjellander · 9 years ago
  10. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  11. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  12. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  13. 381b421 Ping backup connection at a slower rate by Honghai Zhang · 9 years ago
  14. 45b0efd Stop sending stun binding requests after certain amount of time. by honghaiz · 9 years ago
  15. 9e1b992 Clear old decoders after recreating the receiver. by Peter Boström · 9 years ago
  16. 97f7e13 rtcp::ReceiverReport moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  17. 7c704b8 Use webrtc/base/logging.h in stefan@'s ownership. by Peter Boström · 9 years ago
  18. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  19. fcdcf4a Disable RtcEventLogTest.DropOldEvents on DrMemory. by Peter Boström · 9 years ago
  20. 66f7f4e Roll chromium_revision d3aa9b1..626eecf (362950:363027) by kjellander · 9 years ago
  21. fd59523 Add webrtc/base to deprecated APIs. by kjellander · 9 years ago
  22. bc32ab4 Remove 'video_engine_core_unittests' binary. by Peter Boström · 9 years ago
  23. ff24c04 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. by Åsa Persson · 9 years ago
  24. 1a5cf6e Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine). by Fredrik Solenberg · 9 years ago
  25. f7c5776 Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket. by Erik Språng · 9 years ago
  26. 9cf0c3d Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient. by Ivo Creusen · 9 years ago
  27. d048aa0 Make the audio codecs' GN targets self-sufficient by Henrik Lundin · 9 years ago
  28. b4a1ae5 Add separate send-side UMA stats for screenshare and video. by sprang · 9 years ago
  29. 29e3003 Bring back baremetal trybots to the default set. by kjellander@webrtc.org · 9 years ago
  30. 5385554 Roll chromium_revision 7461ceb..d3aa9b1 (362933:362950) by kjellander · 9 years ago
  31. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  32. 74a5ffb Roll chromium_revision f068d2f..7461ceb (362762:362933) by kjellander · 9 years ago
  33. 631e134 Rewrote the thread synchronization parts of the test for the locking in APM in response to a locking problem when running in a single-threaded manner. by peah · 9 years ago
  34. 917ba52 autoroll: Update Clang script path. by kjellander · 9 years ago
  35. 53047c9 Add PRESUBMIT check for native API changes. by kjellander · 9 years ago
  36. c3e0fe7 Make it extra safe when deleting a turn entry. by honghaiz · 9 years ago
  37. 7635684 Fix Mac ObjC PeerConnection API compilation. by tkchin · 9 years ago
  38. 9462052 In some rare Android systems ConnectivityManager may be null. by honghaiz · 9 years ago
  39. a448607 Roll chromium_revision a45c85a..f068d2f (362609:362762) by kjellander · 9 years ago
  40. 3c28d0d Disable PeerConnectionEndToEndTest.Call on Mac. by kjellander@webrtc.org · 9 years ago
  41. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  42. ee524f7 Adding Java binding for CreateSender. by deadbeef · 9 years ago
  43. de0fc58 Adding two more debug macros for logging scalar values to file. by peah · 9 years ago
  44. 7e4e01a Add header extension filtering for WebRtcVoiceEngine/MediaChannel. by solenberg · 9 years ago
  45. 2515af2 Removing some unnecessary string manipulation code from VoEBase::GetVersion(). by solenberg · 9 years ago
  46. d20d247 Fix VideoCaptureAndroid, drop frame when switching camera using textures. by perkj · 9 years ago
  47. 226a602 Fix problem when drawing to the Android Media encoder surface. by perkj · 9 years ago
  48. c729032 Resolves issue with multiple calls to audio unit initialization by henrika · 9 years ago
  49. 40455d6 This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase. by perkj · 9 years ago
  50. e338499 Revert of Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. (patchset #18 id:580001 of https://codereview.webrtc.org/1437463002/ ) by asapersson · 9 years ago
  51. 43b4806 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. by asapersson · 9 years ago
  52. 06104b8 Roll chromium_revision eeff895..a45c85a (362465:362609) by kjellander · 9 years ago
  53. 41b0798 Adding CreatePeerConnection method that uses new PC Initialize method. by deadbeef · 9 years ago
  54. 62a91ee Roll chromium_revision 35f35af..eeff895 (362385:362465) by kjellander · 9 years ago
  55. 187db63 Remove VideoReceiveStream deregister of decoders. by Peter Boström · 9 years ago
  56. 04a6bb9 Roll chromium_revision f9fedae..35f35af (362322:362385) by kjellander · 9 years ago
  57. f94abf7 Nuke webrtc/common_video/plane.*. by Peter Boström · 9 years ago
  58. dfbb3a4 Simplify CodecManager::RegisterEncoder() by kwiberg · 9 years ago
  59. 46c9cc0 Provide method for returning certificate expiration time stamp. by Torbjorn Granlund · 9 years ago
  60. ea07373 Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. by Fredrik Solenberg · 9 years ago
  61. 0de97f1 WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest. by hbos · 9 years ago
  62. ec192bd Revert of Add _decoder CHECK to VCMGenericDecoder constructor. (patchset #2 id:20001 of https://codereview.webrtc.org/1485713002/ ) by kjellander · 9 years ago
  63. cb9792e Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M. by perkj · 9 years ago
  64. 9f8d39d Add simple end to end test for video capture and encode using textures. by perkj · 9 years ago
  65. 021282f Roll chromium_revision 47ce5fe..f9fedae (362117:362322) by kjellander · 9 years ago
  66. 14f4144 Add helper KeepRefUntilDone. by perkj · 9 years ago
  67. ee69ed5 Add separate event for camera freeze. by glaznev · 9 years ago
  68. 70c0e29 Disable PeerConnectionEndToEndTest.Call for TSan. by kjellander@webrtc.org · 9 years ago
  69. f893df3 Add third_party/libc++static to .gitignore by kjellander@webrtc.org · 9 years ago
  70. a443ec1 Add _decoder CHECK to VCMGenericDecoder constructor. by Peter Boström · 9 years ago
  71. 7640ffa Initialize type_preference_ in TestPort. by pbos · 9 years ago
  72. f9203c6 Roll chromium_revision faa24ae..47ce5fe (362083:362117) by kjellander · 9 years ago
  73. 99f8566 Roll chromium_revision 0da9346..faa24ae (362069:362083) by kjellander · 9 years ago
  74. bdf001a Roll chromium_revision 8f57310..0da9346 (362067:362069) by kjellander · 9 years ago
  75. 90728b9 Roll chromium_revision 3e15d1a..8f57310 (362064:362067) by kjellander · 9 years ago
  76. 4c14254 Roll chromium_revision df4d569..3e15d1a (362055:362064) by kjellander · 9 years ago
  77. df3efa8 Introduced the new locking scheme by peah · 9 years ago
  78. 3236b91 Roll chromium_revision c54812d..df4d569 (362052:362055) by kjellander · 9 years ago
  79. 535727e Roll chromium_revision 5ac8f02..c54812d (362046:362052) by kjellander · 9 years ago
  80. ae54b83 Android SurfaceViewRenderer: Add resetStatistics() method by magjed · 9 years ago
  81. 43f1809 Roll chromium_revision 7b99051..5ac8f02 (361977:362046) by kjellander · 9 years ago
  82. 2fe1cb0 Don't overwrite audio stats when they're not available. by andrew · 9 years ago
  83. 7e43138 -Removed the state as an input to the FilterAdaptation function. by peah · 9 years ago
  84. 19822d6 audio_coding: Cleanup duplicated headers after "main" removal. by kjellander · 9 years ago
  85. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  86. ad85622 Use webrtc/base/logging.h for voice_engine. by pbos · 9 years ago
  87. def5820 Default to LS_INFO logging for release builds. by Peter Boström · 9 years ago
  88. 521af4e Remove duplicate decoders in BitrateEstimatorTest. by Peter Boström · 9 years ago
  89. 395c7c6 Re-add missing return in RegisterExternalDecoder. by Peter Boström · 9 years ago
  90. f8385ad rtcp::Pli moved into own file and got a Parse function by danilchap · 9 years ago
  91. e997a7d Call InitDecode with proper resolution. by Peter Boström · 9 years ago
  92. 795dbe4 Remove RegisterExternal{De,En}coder error codes. by Peter Boström · 9 years ago
  93. 34873b5 Roll chromium_revision 7ec1eb8..7b99051 (361868:361977) by kjellander · 9 years ago
  94. 26c8c91 Using Rent-A-Codec for static Codec access in WVoE/MC. by solenberg · 9 years ago
  95. 8779a77 Fix standalone denoiser Android GN compile failure by Magnus Jedvert · 9 years ago
  96. 81b9bfe Added a threadchecking scheme to APM that checks that the APM API calls are called from the correct threads. The actual threadcheckers were, however, removed and will be reintroduced in another upcoming CL. by peah · 9 years ago
  97. 64c0a0a Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ ) by stefan · 9 years ago
  98. 42f580e Leaving all original files in talk/app/webrtc/objc until we can officially tell clients about the new locations. by tkchin · 9 years ago
  99. b1ac203 Introduce helper class NtpTime by danilchap · 9 years ago
  100. 6e40c09 Fix root_files WATCHLIST. by andrew · 9 years ago