Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 7623ce4aeb9130c937ba5836490cbb3a35679e79
  1. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  2. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  3. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  4. d02b0fa Add oldest rotation type option to RTCFileLogger by haysc · 9 years ago
  5. 5e465c3 Make NoiseSuppression not a processing component (bit exact). by solenberg · 9 years ago
  6. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  7. 2d63680 Roll chromium_revision 9dfb3a1..dad6346 (363718:363782) by kjellander · 9 years ago
  8. 7b2f762 Don't call SetPreviewFormat if capturing to textures. by perkj · 9 years ago
  9. edd8fef Add new view that renders local video using AVCaptureLayerPreview. by haysc · 9 years ago
  10. 70f9903 Make HighPassFilter not a ProcessingComponent anymore (bit exact). by solenberg · 9 years ago
  11. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  12. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  13. e10c82d Deletes temporary files that are generated in several ACM unittests. by ivoc · 9 years ago
  14. d7b7ae8 Add encode/decode time tracing to audio_coding. by Peter Boström · 9 years ago
  15. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  16. cd6f539 Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ ) by hbos · 9 years ago
  17. fe32a76 Create fuzzer tests for audio decoders by Henrik Lundin · 9 years ago
  18. ffea13c PRESUBMIT: change native API check from warning to information. by kjellander · 9 years ago
  19. 20ef654 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime(). by hbos · 9 years ago
  20. 325b345 There was an old scaling for CNG 48 kHz in the code, from the time where Audio Coding Module didn't have full 48 kHz support. This CL removes the scaling. by Tina le Grand · 9 years ago
  21. 88eeac4 Adding video_processing to presubmit lint check by mflodman · 9 years ago
  22. 4654d20 Add test which verifies that the RTP header extensions are set correctly for FEC packets. by Stefan Holmer · 9 years ago
  23. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  24. 99ab944 Clang format of video_processing folder. by mflodman · 9 years ago
  25. a440c6f Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718) by kjellander · 9 years ago
  26. 3868692 Free SCTP data channels asynchronously in PeerConnection. by deadbeef · 9 years ago
  27. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  28. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  29. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  30. 0f490a5 Add logs when stun or turn host lookup is completed. by Honghai Zhang · 9 years ago
  31. cd4003f Use @webrtc.org addresses for OWNERS. by Peter Boström · 9 years ago
  32. cf890bc Roll gtest-parallel. by Peter Boström · 9 years ago
  33. 0608dc5 Roll chromium_revision 4918765..3b8be21 (363393:363445) by kjellander · 9 years ago
  34. 5f6deaf Remove unused RTP-header parser. by Peter Boström · 9 years ago
  35. ab82cbb Disable RtcEventLogTest.DropOldEvents on memcheck. by Peter Boström · 9 years ago
  36. 03671cb Use existing parser in ReceivesAndRetransmitsNack. by Peter Boström · 9 years ago
  37. fc47ed6 rtcp::Rrtr block moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  38. 1aa420b Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead. by asapersson · 9 years ago
  39. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  40. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  41. 03f80eb Refactor EglBase configuration. by nisse · 9 years ago
  42. a856542 Initial VideoProcessing refactoring. by mflodman · 9 years ago
  43. 2512f44 Roll chromium_revision 292ab9f..4918765 (363376:363393) by kjellander · 9 years ago
  44. c9f1cb8 Roll chromium_revision 72c3265..292ab9f (363365:363376) by kjellander · 9 years ago
  45. 34a7054 Roll chromium_revision 626eecf..72c3265 (363027:363365) by kjellander · 9 years ago
  46. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  47. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  48. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  49. 381b421 Ping backup connection at a slower rate by Honghai Zhang · 9 years ago
  50. 45b0efd Stop sending stun binding requests after certain amount of time. by honghaiz · 9 years ago
  51. 9e1b992 Clear old decoders after recreating the receiver. by Peter Boström · 9 years ago
  52. 97f7e13 rtcp::ReceiverReport moved into own file and got Parse function by Danil Chapovalov · 9 years ago
  53. 7c704b8 Use webrtc/base/logging.h in stefan@'s ownership. by Peter Boström · 9 years ago
  54. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  55. fcdcf4a Disable RtcEventLogTest.DropOldEvents on DrMemory. by Peter Boström · 9 years ago
  56. 66f7f4e Roll chromium_revision d3aa9b1..626eecf (362950:363027) by kjellander · 9 years ago
  57. fd59523 Add webrtc/base to deprecated APIs. by kjellander · 9 years ago
  58. bc32ab4 Remove 'video_engine_core_unittests' binary. by Peter Boström · 9 years ago
  59. ff24c04 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. by Åsa Persson · 9 years ago
  60. 1a5cf6e Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine). by Fredrik Solenberg · 9 years ago
  61. 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
  62. 9cf0c3d Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient. by Ivo Creusen · 9 years ago
  63. d048aa0 Make the audio codecs' GN targets self-sufficient by Henrik Lundin · 9 years ago
  64. b4a1ae5 Add separate send-side UMA stats for screenshare and video. by sprang · 9 years ago
  65. 29e3003 Bring back baremetal trybots to the default set. by kjellander@webrtc.org · 9 years ago
  66. 5385554 Roll chromium_revision 7461ceb..d3aa9b1 (362933:362950) by kjellander · 9 years ago
  67. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  68. 74a5ffb Roll chromium_revision f068d2f..7461ceb (362762:362933) by kjellander · 9 years ago
  69. 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
  70. 917ba52 autoroll: Update Clang script path. by kjellander · 9 years ago
  71. 53047c9 Add PRESUBMIT check for native API changes. by kjellander · 9 years ago
  72. c3e0fe7 Make it extra safe when deleting a turn entry. by honghaiz · 9 years ago
  73. 7635684 Fix Mac ObjC PeerConnection API compilation. by tkchin · 9 years ago
  74. 9462052 In some rare Android systems ConnectivityManager may be null. by honghaiz · 9 years ago
  75. a448607 Roll chromium_revision a45c85a..f068d2f (362609:362762) by kjellander · 9 years ago
  76. 3c28d0d Disable PeerConnectionEndToEndTest.Call on Mac. by kjellander@webrtc.org · 9 years ago
  77. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  78. ee524f7 Adding Java binding for CreateSender. by deadbeef · 9 years ago
  79. de0fc58 Adding two more debug macros for logging scalar values to file. by peah · 9 years ago
  80. 7e4e01a Add header extension filtering for WebRtcVoiceEngine/MediaChannel. by solenberg · 9 years ago
  81. 2515af2 Removing some unnecessary string manipulation code from VoEBase::GetVersion(). by solenberg · 9 years ago
  82. d20d247 Fix VideoCaptureAndroid, drop frame when switching camera using textures. by perkj · 9 years ago
  83. 226a602 Fix problem when drawing to the Android Media encoder surface. by perkj · 9 years ago
  84. c729032 Resolves issue with multiple calls to audio unit initialization by henrika · 9 years ago
  85. 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
  86. 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
  87. 43b4806 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. by asapersson · 9 years ago
  88. 06104b8 Roll chromium_revision eeff895..a45c85a (362465:362609) by kjellander · 9 years ago
  89. 41b0798 Adding CreatePeerConnection method that uses new PC Initialize method. by deadbeef · 9 years ago
  90. 62a91ee Roll chromium_revision 35f35af..eeff895 (362385:362465) by kjellander · 9 years ago
  91. 187db63 Remove VideoReceiveStream deregister of decoders. by Peter Boström · 9 years ago
  92. 04a6bb9 Roll chromium_revision f9fedae..35f35af (362322:362385) by kjellander · 9 years ago
  93. f94abf7 Nuke webrtc/common_video/plane.*. by Peter Boström · 9 years ago
  94. dfbb3a4 Simplify CodecManager::RegisterEncoder() by kwiberg · 9 years ago
  95. 46c9cc0 Provide method for returning certificate expiration time stamp. by Torbjorn Granlund · 9 years ago
  96. ea07373 Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. by Fredrik Solenberg · 9 years ago
  97. 0de97f1 WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest. by hbos · 9 years ago
  98. ec192bd Revert of Add _decoder CHECK to VCMGenericDecoder constructor. (patchset #2 id:20001 of https://codereview.webrtc.org/1485713002/ ) by kjellander · 9 years ago
  99. cb9792e Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M. by perkj · 9 years ago
  100. 9f8d39d Add simple end to end test for video capture and encode using textures. by perkj · 9 years ago