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webrtc / src / webrtc / c40089392ccf8898607c3ea46d2776031b7128ea
  1. c400893 Add DesktopRectTest for UnionWith() function by zijiehe · 8 years ago
  2. eaabbe0 Fixing memory leak of generated session descriptions on Android. by deadbeef · 8 years ago
  3. 47be53a Dont request keyframes if the stream is inactive or if we are currently receiving a keyframe. by philipel · 8 years ago
  4. cfc8721 Revert of Removes usage of native base::android::GetApplicationContext() (patchset #6 id:120001 of https://codereview.webrtc.org/2888093004/ ) by sakal · 8 years ago
  5. 98eef6c Removes usage of native base::android::GetApplicationContext() by sakal · 8 years ago
  6. f9ccc94 Delete FilesystemInterface::DeleteFolderAndContents and related methods. by nisse · 8 years ago
  7. 72aaf0a Reland of reduce dependencies on rtc::FileSystem in FileRotatingStream tests... (patchset #1 id:1 of https://codereview.webrtc.org/2885393002/ ) by nisse · 8 years ago
  8. 15222f4 Break backwards traversal loop if we have looped around all packets in the PacketBuffer for H264 frames. by philipel · 8 years ago
  9. 7b2962c iOS: Fix runtime error in AppRTCMobile by hewwatt · 8 years ago
  10. 297bd8c Add methods to change enabled events in PhysicalSocket. by jbauch · 8 years ago
  11. ccb281e Updated comments for unit tests to validate iOS audio session isInterrupted flag does not get reset correctly. by jtteh · 8 years ago
  12. 82158aa Simple tests for Call::SetBitrateConfig. by zstein · 8 years ago
  13. 953e232 Remove gflags dependency for screenshare_loopback by kjellander · 8 years ago
  14. 35f747a Add log message to help analyze why echo likelihood > 1.1 by ivoc · 8 years ago
  15. cfe92cd Don't add FEC and RTX overheads when calculating a padding packet's maximum payload size. by erikvarga · 8 years ago
  16. bec3fa9 Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ ) by kthelgason · 8 years ago
  17. d5acef3 Revert of Reduce dependencies on rtc::FileSystem in FileRotatingStream tests, adding helpers in webrtc::test:: (patchset #7 id:120001 of https://codereview.webrtc.org/2872283002/ ) by ehmaldonado · 8 years ago
  18. 65aad01 Remove hardcoded kValueSizeBytes values from variable-length header extensions. by erikvarga · 8 years ago
  19. 8357b2c Remove temporary include of builtin_audio_encoder_factory.h. by ossu · 8 years ago
  20. 2fdddbf Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890513002/ ) by kthelgason · 8 years ago
  21. 373abef Reduce dependencies on rtc::FileSystem in FileRotatingStream tests. by nisse · 8 years ago
  22. 92b90e4 Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 8 years ago
  23. 6cfa9f1 Add a DesktopRect::UnionWith() function to extend current instance to cover both instances by zijiehe · 8 years ago
  24. 8e98047 Revert of Split iOS sdk in to separate targets (patchset #13 id:280001 of https://codereview.webrtc.org/2862543002/ ) by charujain · 8 years ago
  25. 22e0182 Request keyframe if the first received frame is not a keyframe. by philipel · 8 years ago
  26. 04625df Removed implicit divisions in the residual echo detector by peah · 8 years ago
  27. 6003fb9 Fixed NetEq overflow bug. by ivoc · 8 years ago
  28. 45f966d Split iOS sdk in to separate targets by kthelgason · 8 years ago
  29. b89b34c Adds fuzzer for the residual echo detector. by ivoc · 8 years ago
  30. 7c6ea70 Ensures the residual echo detector does not requiring band-splitting by peah · 8 years ago
  31. 74690e9 Delete unused features of AsyncInvoke. by nisse · 8 years ago
  32. 5b2b33a Update testing tools (AppRTC, Go) to new versions by oprypin · 8 years ago
  33. 78bbf3a New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 8 years ago
  34. eca698b Add Parser to analyse the results of the network tester. by michaelt · 8 years ago
  35. 78dd13d Corrected the number of channels used when AEC3 is run on stereo input. by peah · 8 years ago
  36. 16c2426 Remove gflags dependency for event_log_visualizer and activity_metric by kjellander · 8 years ago
  37. 26c636b Update adaptation stats to support degradations in both resolution and framerate. by asapersson · 8 years ago
  38. 61de4a2 Get tests working on systems that only support IPv6. by deadbeef · 8 years ago
  39. f0995a9 Adding target to track asm_defines.h by mbonadei · 8 years ago
  40. e855309 Moving compile_assert_c.h to webrtc/base by mbonadei · 8 years ago
  41. b2b2b81 Moving the residual echo detector outside of band-scheme in APM by peah · 8 years ago
  42. a24bcce Move webrtc/video_frame to common_video/include. by nisse · 8 years ago
  43. 9d4241c Add a couple of checks to FrameBuffer while we're continuing to look at RtpFrameReferenceFinder. by tommi · 8 years ago
  44. 0e31205 Initialize PeerConnection members in declaration order and destroy them in reverse order. by terelius · 8 years ago
  45. 290628c Run clang-format on sigslot.h. by deadbeef · 8 years ago
  46. 4fd55b0 Relanding #2: Fixing crash that can occur if signal is modified while firing. by deadbeef · 8 years ago
  47. 5dee2d1 Move test-only code to GN target rtc_base_test_utils. by nisse · 8 years ago
  48. 7aa35a5 Support running AppRTC without a TURN ICE server by oprypin · 8 years ago
  49. 0d1e81b WebRtcVideoEncoderFactory cleanup by magjed · 8 years ago
  50. 7abae18 Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ ) by zhihuang · 8 years ago
  51. 8a2a0a0 Revert of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2873803002/ ) by zhihuang · 8 years ago
  52. acce975 An example of Unity native plugin of webrtc for Windows OS by gyzhou · 8 years ago
  53. 71f49e5 Make AudioSinkInterface::Data hold a const pointer to the audio buffer. by yujo · 8 years ago
  54. ba6f478 Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket. by nisse · 8 years ago
  55. 55621d4 Add untracked header files to GN targets in audio_coding by henrik.lundin · 8 years ago
  56. 654261a DCHECK that we don't insert nullptr into event log. by terelius · 8 years ago
  57. 6a7a94e Add some unit tests to ReceiveStatsticsProxy. by asapersson · 8 years ago
  58. 03655e2 Add support for I444 in VideoFrameBuffer by magjed · 8 years ago
  59. fc4e664 Do not delete temporary directory if user specified it manually. by sakal · 8 years ago
  60. 49f9b64 Add write support for the RtpStreamId and RepairedRtpStreamId header extensions. by erikvarga · 8 years ago
  61. 861eaa3 Updating VCM owners to reflect current active persons in the project. by mflodman · 8 years ago
  62. 7b4b542 Disable the residual echo detector in audio mixer. by aleloi · 8 years ago
  63. f6184e4 Fix audio device excessive logging on Windows by lliuu · 8 years ago
  64. 4acd8c1 Configured VCMTiming with sender defining delay times. by gnish · 8 years ago
  65. cd2f080 Add untracked headers in modules/rtp_rtcp by danilchap · 8 years ago
  66. 5b82269 Handle padded audio packets correctly by henrik.lundin · 8 years ago
  67. 5a51189 Delete left-over declaration of AdjustCurrentProcessPrivilege. by nisse · 8 years ago
  68. d1df7af Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 8 years ago
  69. b2a8855 Fixing video loopback test with encoder factory. by minyue · 8 years ago
  70. f4e86ab Deleted unused method EstimateMTU, and the WinPing class. by nisse · 8 years ago
  71. 77717f6 iOS audio session isInterrupted flag does not get reset correctly: by jtteh · 8 years ago
  72. 10a054d Delete helper class MediaTypePacketReceiver. by nisse · 8 years ago
  73. 4647b14 Drop deprecated AudioFrameOperations::Scale method signatures by oprypin · 8 years ago
  74. a1a924a Test for Gradle project generation. by sakal · 8 years ago
  75. 7118528 Moving scripts to download and build apprtc/collider. by mbonadei · 8 years ago
  76. eda1919 Rename tools-webrtc -> tools_webrtc by Henrik Kjellander · 8 years ago
  77. 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
  78. e334ec8 Fixing invalid IPv6 address parsing stack underflow on Windows. by deadbeef · 8 years ago
  79. cfc8c13 Make WebRtcAudioEffects and its create method public. by mellem · 8 years ago
  80. d2cafce Delete unused class SharedExclusiveLock. by nisse · 8 years ago
  81. 6b17ee2 Don't add or rename files in webrtc/ and webrtc/api/ without a proper review by kwiberg · 8 years ago
  82. df1eb18 Actually move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/ by magjed · 8 years ago
  83. 59a36ed Delete method MessageQueue::set_socketserver by nisse · 8 years ago
  84. e6bd325 Update comments for removal of MediaController. by nisse · 8 years ago
  85. ed69f12 Remove deprecated API by ilnik · 8 years ago
  86. e37a700 Refactor TestClient to use std::unique_ptr, and fix VirtualSocketServerTest leaks. by nisse · 8 years ago
  87. 886a9f8 Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver. by deadbeef · 8 years ago
  88. 1641e82 Add myself as OWNER of webrtc/api/ and webrtc/base/ by kwiberg · 8 years ago
  89. 47e6d13 Create an OWNERS file in webrtc/api/audio_codecs/ by kwiberg · 8 years ago
  90. 7f952f3 Fix webrtcsdp_unittest. by ehmaldonado · 8 years ago
  91. db9aaa5 When a data channel fails to be created, return nil instead of crashing. by deadbeef · 8 years ago
  92. ca384b9 Prevent residual echo likelihood values greater than 1.0. by ivoc · 8 years ago
  93. de184ce NetEq: Fix a bug in expand_rate and speech_expand_rate calculation by henrik.lundin · 8 years ago
  94. c0ff88b Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  95. c63e36e Blacklisting of HW-AEC/NS and OpenSL must now be done by the WebRTC client. by henrika · 8 years ago
  96. 57a4d1c Add support for media recorders in Camera1Capturer. by sakal · 8 years ago
  97. d6b3a36 Make fps NSInteger in startCaptureWithDevice. by sakal · 8 years ago
  98. 6038d77 Fixing pseudotcp_parser_fuzzer crash with NO_MAIN_THREAD_WRAPPING. by deadbeef · 8 years ago
  99. 211c500 Don't initiate perodic probing if we don't have a bandwidth estimate. by philipel · 8 years ago
  100. 83935da Remove layer_sync from TL frame config. by pbos · 8 years ago