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gerrit-public.fairphone.software / platform / external / webrtc / 26679d6d90aeb7114b6bcc86b26921768256e203
  1. 26679d6 ViEFrameCallback::DeliverFrame: Make I420VideoFrame const ref. by Magnus Jedvert · 10 years ago
  2. 3211934 Fix build breakage in WrappedI420Buffer::native_handle() by Per · 10 years ago
  3. 75db861 Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection. by Per · 10 years ago
  4. e1c1ee2 EncodedVideoData is unused, so remove it by Karl Wiberg · 10 years ago
  5. e095148 Port some fixes in AppRTCDemo. by Alex Glaznev · 10 years ago
  6. be508a1 Implement Tcp Reconnect for TCPPort. by Guo-wei Shieh · 10 years ago
  7. ef88309 Cleanup: Forward declare AudioFrame type in voiceprocess.h by Thiago Farina · 10 years ago
  8. ae0f0ee Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro. by Thiago Farina · 10 years ago
  9. 7351f46 Don't send STUN pings if we don't have a remote ufrag and pwd. by Peter Thatcher · 10 years ago
  10. bc4b934 Add a DCHECK to RegisterModule to make sure it's called on the controller thread. by Tommi · 10 years ago
  11. 7f375f0 ProcessThreadImpl - hold the lock while checking thread_ and calling ProcessThreadAttached(). by Tommi · 10 years ago
  12. 3354419 Zero copy AndroidVideeCapturer. by Per · 10 years ago
  13. 037bad7 ~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug. by Henrik Boström · 10 years ago
  14. cb76b89 Cleanup: Move json.h into rtc namespace. by Thiago Farina · 10 years ago
  15. 0dd5802 Update callers to include messagedigest.h. by Thiago Farina · 10 years ago
  16. db313b6 Disable EndToEndTest.ReceivedFecPacketsNotNacked on all platforms. by Henrik Kjellander · 10 years ago
  17. d4e7501 Refactor audio_coding/codecs/isac/fix: Removed usage of trivial macro WEBRTC_SPL_LSHIFT_W32() by Bjorn Volcker · 10 years ago
  18. 64c1e8c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  19. aaf61e4 Cleanup: Remove MD5_CTX typedef. by Thiago Farina · 10 years ago
  20. fa16dda Revert "Port frame_analyzer and rgba_to_i420_converter targets to GN build." by Henrik Kjellander · 10 years ago
  21. 6ac53b2 Port frame_analyzer and rgba_to_i420_converter targets to GN build. by Henrik Kjellander · 10 years ago
  22. 722ef1f Remove henrike@ from OWNERS by Henrik Kjellander · 10 years ago
  23. cf3c83e Revert "Split EventWrapper in twain." by Minyue · 10 years ago
  24. 31331cf Revert "Enable CVO by default through webrtc pipeline." by Minyue · 10 years ago
  25. d91cb5d Reduce the number of Chromium dependencies synced. by Henrik Kjellander · 10 years ago
  26. 3cd9eaf Ensures that AudioManager.isVolumeFixed() is only used for Android L and above by henrika · 10 years ago
  27. f536a50 Remove duplicated source listing of gtest_prod_util.h by Henrik Kjellander · 10 years ago
  28. f809b9b Fix bug in WebRtcIsacfix_FilterMaLoopNeon. by Zhongwei Yao · 10 years ago
  29. 9cb1f30 Remove er_tables_xor.h. by Peter Boström · 10 years ago
  30. 1b1c15c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  31. 4b3c0d6 Use WebRTC API to convert byteorder in srtpfilter. by Jiayang Liu · 10 years ago
  32. 4825356 RTCDataChannel: Unregister data channel observer on dealloc. by Zeke Chin · 10 years ago
  33. 379069f VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const. by Magnus Jedvert · 10 years ago
  34. 0828a0c Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 10 years ago
  35. 23914fe Reject RTP one-byte extension ID 0. by Peter Boström · 10 years ago
  36. 903c0f2 Avoid critsect for protection- and qm setting callbacks in VideoSender. by mflodman · 10 years ago
  37. 738a5b4 Remove old suppression for ProcessThreadImpl. by Tommi · 10 years ago
  38. bc46bf2 common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM by Bjorn Volcker · 10 years ago
  39. 0194d32 Add WebRtcAudioManager to peerconnection_jar library by Alex Glaznev · 10 years ago
  40. 65f74a1 Revert "Suppress data races in libjingle_peerconnection_unittest" by Tommi · 10 years ago
  41. 2c9c83d Remove non-functional asynchronous resampling mode. by Andrew MacDonald · 10 years ago
  42. 45c6449 Introduce CodecManager and move code from AudioCodingModuleImpl by Henrik Lundin · 10 years ago
  43. f7b9cf5 Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan by Minyue Li · 10 years ago
  44. 842a4a6 Add locks to Start(), Stop() methods in ProcessThread. by Tommi · 10 years ago
  45. 22e209d Introduce AudioCodingModuleImpl::current_encoder_ by Henrik Lundin · 10 years ago
  46. 582f80e Clamp decoder sample rate to 32000 in iSAC by Henrik Lundin · 10 years ago
  47. 1ecfd55 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)' by Magnus Jedvert · 10 years ago
  48. 451b614 Fix gyp path for bwe simulator include. by Stefan Holmer · 10 years ago
  49. 8e9c67e Suppress data races in libjingle_peerconnection_unittest by Henrik Kjellander · 10 years ago
  50. 9f52448 Roll chromium_revision 4d63ee8..719b839 (322012:322539) by Henrik Kjellander · 10 years ago
  51. 6b3ccfc GN: Cleanup no longer needed libvpx config. by Henrik Kjellander · 10 years ago
  52. 819011c Additional suppression for TSan deadlock detection by Henrik Kjellander · 10 years ago
  53. dfd53fe Raise streams for SetMaxSendBitrates above 2000k. by Peter Boström · 10 years ago
  54. 53eda3d Add tests for r8811. by Peter Boström · 10 years ago
  55. b3fc48b Update the notice about the slow Chromium sync. by Henrik Kjellander · 10 years ago
  56. 1d36003 Suppress TSan errors triggered when deadlock detection is enabled. by Henrik Kjellander · 10 years ago
  57. 9ff73f5 Final minor fix in WebRtcAudioManager by henrika · 10 years ago
  58. 424694c audio_processing/agc: Put entire method set_output_will_be_muted() under lock by Bjorn Volcker · 10 years ago
  59. 75a0255 Handle borked Android cameras gracefully. by Per · 10 years ago
  60. 8324b52 Adding playout volume control to WebRtcAudioTrack.java. by henrika · 10 years ago
  61. 8ed6a4b Remove unused non-standard capture stats. by Peter Boström · 10 years ago
  62. 3954e1d Remove unused implementations in cricket::VideoFrame by Magnus Jedvert · 10 years ago
  63. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 10 years ago
  64. bef8d2d Add a lock to NSSContext to fix data race by Jiayang Liu · 10 years ago
  65. b8cfa68 Update speed setting in VP9. by Marco · 10 years ago
  66. 74d9ed7 Report send codec name in GetStats(). by Peter Boström · 10 years ago
  67. d6f4c25 Reject streams reusing simulcast or RTX SSRCs. by Peter Boström · 10 years ago
  68. a990784 AcmReceiver: index decoders by payload type instead of ACM codec ID by Jelena Marusic · 10 years ago
  69. 9b5f96e Add some sanity CHECKs to webrtc::Call. by Peter Boström · 10 years ago
  70. c79f7ed Fix build error introduced by r8864. by Stefan Holmer · 10 years ago
  71. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  72. 5225dd8 If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size. by Brave Yao · 10 years ago
  73. dfa3605 Reparent Nonlinear beamformer under beamforming interface. by Michael Graczyk · 10 years ago
  74. bf395c1 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 10 years ago
  75. caae5d4 Bye request should use POST not GET by Chuck Hays · 10 years ago
  76. 190c3ca Register sample rate of Audio RED in RTPPayloadRegistry. by Minyue Li · 10 years ago
  77. 79064e5 Fix crash on decode found by fuzz tester. by Stefan Holmer · 10 years ago
  78. 3fbf99c Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
  79. 855acf7 Remove video from WebRTC Android example. by Per · 10 years ago
  80. d4362cd Reject StreamParams with RTX SSRCs not in ssrcs. by Peter Boström · 10 years ago
  81. a49f515 Roll chromium_revision da9a1c0..4d63ee8 (321718:322012) by Henrik Kjellander · 10 years ago
  82. 1ccd8b4 Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
  83. 245989b Address comments from cr 43769004. by Tommi · 10 years ago
  84. 0e209b0 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 10 years ago
  85. e61c64d Delete NullVideoRenderer by Magnus Jedvert · 10 years ago
  86. 07a4ba5 Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them. by Niklas Enbom · 10 years ago
  87. ac27e20 Delete VideoAdapter::AdaptFrame by Magnus Jedvert · 10 years ago
  88. 45636ec Post Git switch: Update codereview.settings and remove drover.properties by Henrik Kjellander · 10 years ago
  89. 68a5418 Enable PENDING_REF_PREFIX in codereview.settings. by Henrik Kjellander · 10 years ago
  90. 4d14592 rtc::Buffer: Restore length method for backwards compatibility by kwiberg@webrtc.org · 10 years ago
  91. deafa7b Remove I420VideoFrame::SwapFrame by magjed@webrtc.org · 10 years ago
  92. 2d2a30c Remove I420VideoFrame::CloneFrame by magjed@webrtc.org · 10 years ago
  93. 0b52ceb Improve logging and add DCHECKs in codec database. by pbos@webrtc.org · 10 years ago
  94. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
  95. e815290 Update README instructions for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  96. a5f6fb5 Permit single-stream max bitrates above 2000k. by pbos@webrtc.org · 10 years ago
  97. a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 10 years ago
  98. a3ffc56 Allow setting thread priorities in Chromium on all but linux platforms. by tommi@webrtc.org · 10 years ago
  99. 39fc1d3 Disable PeerConnectionClientTest.testLoopbackVp9 by henrik.lundin@webrtc.org · 10 years ago
  100. 0b44b58 Limit disabling of PeerConnectionEndToEndTest.Call to Windows by henrik.lundin@webrtc.org · 10 years ago