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gerrit-public.fairphone.software / platform / external / webrtc / 4b3c0d6f34c379c2a06f654b62403876b20180c6
  1. 4b3c0d6 Use WebRTC API to convert byteorder in srtpfilter. by Jiayang Liu · 9 years ago
  2. 4825356 RTCDataChannel: Unregister data channel observer on dealloc. by Zeke Chin · 9 years ago
  3. 379069f VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const. by Magnus Jedvert · 9 years ago
  4. 0828a0c Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 9 years ago
  5. 23914fe Reject RTP one-byte extension ID 0. by Peter Boström · 9 years ago
  6. 903c0f2 Avoid critsect for protection- and qm setting callbacks in VideoSender. by mflodman · 9 years ago
  7. 738a5b4 Remove old suppression for ProcessThreadImpl. by Tommi · 9 years ago
  8. bc46bf2 common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM by Bjorn Volcker · 9 years ago
  9. 0194d32 Add WebRtcAudioManager to peerconnection_jar library by Alex Glaznev · 9 years ago
  10. 65f74a1 Revert "Suppress data races in libjingle_peerconnection_unittest" by Tommi · 9 years ago
  11. 2c9c83d Remove non-functional asynchronous resampling mode. by Andrew MacDonald · 9 years ago
  12. 45c6449 Introduce CodecManager and move code from AudioCodingModuleImpl by Henrik Lundin · 9 years ago
  13. f7b9cf5 Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan by Minyue Li · 9 years ago
  14. 842a4a6 Add locks to Start(), Stop() methods in ProcessThread. by Tommi · 9 years ago
  15. 22e209d Introduce AudioCodingModuleImpl::current_encoder_ by Henrik Lundin · 9 years ago
  16. 582f80e Clamp decoder sample rate to 32000 in iSAC by Henrik Lundin · 9 years ago
  17. 1ecfd55 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)' by Magnus Jedvert · 9 years ago
  18. 451b614 Fix gyp path for bwe simulator include. by Stefan Holmer · 9 years ago
  19. 8e9c67e Suppress data races in libjingle_peerconnection_unittest by Henrik Kjellander · 9 years ago
  20. 9f52448 Roll chromium_revision 4d63ee8..719b839 (322012:322539) by Henrik Kjellander · 9 years ago
  21. 6b3ccfc GN: Cleanup no longer needed libvpx config. by Henrik Kjellander · 9 years ago
  22. 819011c Additional suppression for TSan deadlock detection by Henrik Kjellander · 9 years ago
  23. dfd53fe Raise streams for SetMaxSendBitrates above 2000k. by Peter Boström · 9 years ago
  24. 53eda3d Add tests for r8811. by Peter Boström · 9 years ago
  25. b3fc48b Update the notice about the slow Chromium sync. by Henrik Kjellander · 9 years ago
  26. 1d36003 Suppress TSan errors triggered when deadlock detection is enabled. by Henrik Kjellander · 9 years ago
  27. 9ff73f5 Final minor fix in WebRtcAudioManager by henrika · 9 years ago
  28. 424694c audio_processing/agc: Put entire method set_output_will_be_muted() under lock by Bjorn Volcker · 9 years ago
  29. 75a0255 Handle borked Android cameras gracefully. by Per · 9 years ago
  30. 8324b52 Adding playout volume control to WebRtcAudioTrack.java. by henrika · 9 years ago
  31. 8ed6a4b Remove unused non-standard capture stats. by Peter Boström · 9 years ago
  32. 3954e1d Remove unused implementations in cricket::VideoFrame by Magnus Jedvert · 9 years ago
  33. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 9 years ago
  34. bef8d2d Add a lock to NSSContext to fix data race by Jiayang Liu · 9 years ago
  35. b8cfa68 Update speed setting in VP9. by Marco · 9 years ago
  36. 74d9ed7 Report send codec name in GetStats(). by Peter Boström · 9 years ago
  37. d6f4c25 Reject streams reusing simulcast or RTX SSRCs. by Peter Boström · 9 years ago
  38. a990784 AcmReceiver: index decoders by payload type instead of ACM codec ID by Jelena Marusic · 9 years ago
  39. 9b5f96e Add some sanity CHECKs to webrtc::Call. by Peter Boström · 9 years ago
  40. c79f7ed Fix build error introduced by r8864. by Stefan Holmer · 9 years ago
  41. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 9 years ago
  42. 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 · 9 years ago
  43. dfa3605 Reparent Nonlinear beamformer under beamforming interface. by Michael Graczyk · 9 years ago
  44. bf395c1 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 9 years ago
  45. caae5d4 Bye request should use POST not GET by Chuck Hays · 9 years ago
  46. 190c3ca Register sample rate of Audio RED in RTPPayloadRegistry. by Minyue Li · 9 years ago
  47. 79064e5 Fix crash on decode found by fuzz tester. by Stefan Holmer · 9 years ago
  48. 3fbf99c Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 9 years ago
  49. 855acf7 Remove video from WebRTC Android example. by Per · 9 years ago
  50. d4362cd Reject StreamParams with RTX SSRCs not in ssrcs. by Peter Boström · 9 years ago
  51. a49f515 Roll chromium_revision da9a1c0..4d63ee8 (321718:322012) by Henrik Kjellander · 9 years ago
  52. 1ccd8b4 Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 9 years ago
  53. 245989b Address comments from cr 43769004. by Tommi · 9 years ago
  54. 0e209b0 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 9 years ago
  55. e61c64d Delete NullVideoRenderer by Magnus Jedvert · 9 years ago
  56. 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 · 9 years ago
  57. ac27e20 Delete VideoAdapter::AdaptFrame by Magnus Jedvert · 9 years ago
  58. 45636ec Post Git switch: Update codereview.settings and remove drover.properties by Henrik Kjellander · 9 years ago
  59. 68a5418 Enable PENDING_REF_PREFIX in codereview.settings. by Henrik Kjellander · 9 years ago
  60. 4d14592 rtc::Buffer: Restore length method for backwards compatibility by kwiberg@webrtc.org · 9 years ago
  61. deafa7b Remove I420VideoFrame::SwapFrame by magjed@webrtc.org · 9 years ago
  62. 2d2a30c Remove I420VideoFrame::CloneFrame by magjed@webrtc.org · 9 years ago
  63. 0b52ceb Improve logging and add DCHECKs in codec database. by pbos@webrtc.org · 9 years ago
  64. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 9 years ago
  65. e815290 Update README instructions for Android AppRTCDemo. by glaznev@webrtc.org · 9 years ago
  66. a5f6fb5 Permit single-stream max bitrates above 2000k. by pbos@webrtc.org · 9 years ago
  67. a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 9 years ago
  68. a3ffc56 Allow setting thread priorities in Chromium on all but linux platforms. by tommi@webrtc.org · 9 years ago
  69. 39fc1d3 Disable PeerConnectionClientTest.testLoopbackVp9 by henrik.lundin@webrtc.org · 9 years ago
  70. 0b44b58 Limit disabling of PeerConnectionEndToEndTest.Call to Windows by henrik.lundin@webrtc.org · 9 years ago
  71. 64eb2ff iOS library build script by tkchin@webrtc.org · 9 years ago
  72. 9509fbf Split EventWrapper in twain. by tommi@webrtc.org · 9 years ago
  73. 82e8ae4 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest by henrik.lundin@webrtc.org · 9 years ago
  74. 2b4ce3a Convert webrtc/video/ abort/assert to CHECK/DCHECK. by pbos@webrtc.org · 9 years ago
  75. 41d2bef Limit RED audio payload to narrow band. by minyue@webrtc.org · 9 years ago
  76. 1596a4f Temporarily disable SetPriority when building with Chromium. by tommi@webrtc.org · 9 years ago
  77. d4e7d49 Scaler: Recycle allocations using buffer pool. by magjed@webrtc.org · 9 years ago
  78. 09b6ff9 Disable PLC for iSAC by henrik.lundin@webrtc.org · 9 years ago
  79. ee0c5af Remove unused version.py script. by kjellander@webrtc.org · 9 years ago
  80. aa0bbab Fix build failure by jmarusic@webrtc.org · 9 years ago
  81. a4bef3e AcmReceiver: use std::map instead of an array to keep the list of decoders by jmarusic@webrtc.org · 9 years ago
  82. 3335a4f Prevent asserting on unset start bitrate. by pbos@webrtc.org · 9 years ago
  83. 50ed0d9 Roll chromium_revision 6311617..da9a1c0 (321517:321718) by kjellander@webrtc.org · 9 years ago
  84. e5e92bd Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) by kjellander@webrtc.org · 9 years ago
  85. cfde27e Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. by kjellander@webrtc.org · 9 years ago
  86. 38492c5 Re-land 8810 "- Add a SetPriority method to ThreadWr..." by tommi@webrtc.org · 9 years ago
  87. 90a1cb4 Revert 8810 "- Add a SetPriority method to ThreadWrapper" by tommi@webrtc.org · 9 years ago
  88. b789f62 Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." by tommi@webrtc.org · 9 years ago
  89. 0c34001 Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." by tommi@webrtc.org · 9 years ago
  90. 346a64b Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. by braveyao@webrtc.org · 9 years ago
  91. 4553941 Document the 'int' return value of Resampler methods. by wtc@chromium.org · 9 years ago
  92. 3200a64 Minor fix for MIPS Android build. by andrew@webrtc.org · 9 years ago
  93. 4ddc938 Support VP8 hardware encoding and decoding on IA devices. by glaznev@webrtc.org · 9 years ago
  94. b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 9 years ago
  95. b6817d7 - Add a SetPriority method to ThreadWrapper by tommi@webrtc.org · 9 years ago
  96. 66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 9 years ago
  97. 8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 9 years ago
  98. a3209a2 Release buffer pool in Vp8DecoderImpl::Release(). by pbos@webrtc.org · 9 years ago
  99. 8904290 Make screenshare target bitrate experiment always on by pbos@webrtc.org · 9 years ago
  100. d9c5024 Roll chromium_revision bd49b12..6311617 (320783:321517) by kjellander@webrtc.org · 9 years ago