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gerrit-public.fairphone.software / platform / external / webrtc / 8324b525dce2c502bbd24b3946bbae207645cde9
  1. 8324b52 Adding playout volume control to WebRtcAudioTrack.java. by henrika · 10 years ago
  2. 8ed6a4b Remove unused non-standard capture stats. by Peter Boström · 10 years ago
  3. 3954e1d Remove unused implementations in cricket::VideoFrame by Magnus Jedvert · 10 years ago
  4. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 10 years ago
  5. bef8d2d Add a lock to NSSContext to fix data race by Jiayang Liu · 10 years ago
  6. b8cfa68 Update speed setting in VP9. by Marco · 10 years ago
  7. 74d9ed7 Report send codec name in GetStats(). by Peter Boström · 10 years ago
  8. d6f4c25 Reject streams reusing simulcast or RTX SSRCs. by Peter Boström · 10 years ago
  9. a990784 AcmReceiver: index decoders by payload type instead of ACM codec ID by Jelena Marusic · 10 years ago
  10. 9b5f96e Add some sanity CHECKs to webrtc::Call. by Peter Boström · 10 years ago
  11. c79f7ed Fix build error introduced by r8864. by Stefan Holmer · 10 years ago
  12. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  13. 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
  14. dfa3605 Reparent Nonlinear beamformer under beamforming interface. by Michael Graczyk · 10 years ago
  15. bf395c1 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 10 years ago
  16. caae5d4 Bye request should use POST not GET by Chuck Hays · 10 years ago
  17. 190c3ca Register sample rate of Audio RED in RTPPayloadRegistry. by Minyue Li · 10 years ago
  18. 79064e5 Fix crash on decode found by fuzz tester. by Stefan Holmer · 10 years ago
  19. 3fbf99c Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
  20. 855acf7 Remove video from WebRTC Android example. by Per · 10 years ago
  21. d4362cd Reject StreamParams with RTX SSRCs not in ssrcs. by Peter Boström · 10 years ago
  22. a49f515 Roll chromium_revision da9a1c0..4d63ee8 (321718:322012) by Henrik Kjellander · 10 years ago
  23. 1ccd8b4 Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
  24. 245989b Address comments from cr 43769004. by Tommi · 10 years ago
  25. 0e209b0 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 10 years ago
  26. e61c64d Delete NullVideoRenderer by Magnus Jedvert · 10 years ago
  27. 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
  28. ac27e20 Delete VideoAdapter::AdaptFrame by Magnus Jedvert · 10 years ago
  29. 45636ec Post Git switch: Update codereview.settings and remove drover.properties by Henrik Kjellander · 10 years ago
  30. 68a5418 Enable PENDING_REF_PREFIX in codereview.settings. by Henrik Kjellander · 10 years ago
  31. 4d14592 rtc::Buffer: Restore length method for backwards compatibility by kwiberg@webrtc.org · 10 years ago
  32. deafa7b Remove I420VideoFrame::SwapFrame by magjed@webrtc.org · 10 years ago
  33. 2d2a30c Remove I420VideoFrame::CloneFrame by magjed@webrtc.org · 10 years ago
  34. 0b52ceb Improve logging and add DCHECKs in codec database. by pbos@webrtc.org · 10 years ago
  35. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
  36. e815290 Update README instructions for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  37. a5f6fb5 Permit single-stream max bitrates above 2000k. by pbos@webrtc.org · 10 years ago
  38. a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 10 years ago
  39. a3ffc56 Allow setting thread priorities in Chromium on all but linux platforms. by tommi@webrtc.org · 10 years ago
  40. 39fc1d3 Disable PeerConnectionClientTest.testLoopbackVp9 by henrik.lundin@webrtc.org · 10 years ago
  41. 0b44b58 Limit disabling of PeerConnectionEndToEndTest.Call to Windows by henrik.lundin@webrtc.org · 10 years ago
  42. 64eb2ff iOS library build script by tkchin@webrtc.org · 10 years ago
  43. 9509fbf Split EventWrapper in twain. by tommi@webrtc.org · 10 years ago
  44. 82e8ae4 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest by henrik.lundin@webrtc.org · 10 years ago
  45. 2b4ce3a Convert webrtc/video/ abort/assert to CHECK/DCHECK. by pbos@webrtc.org · 10 years ago
  46. 41d2bef Limit RED audio payload to narrow band. by minyue@webrtc.org · 10 years ago
  47. 1596a4f Temporarily disable SetPriority when building with Chromium. by tommi@webrtc.org · 10 years ago
  48. d4e7d49 Scaler: Recycle allocations using buffer pool. by magjed@webrtc.org · 10 years ago
  49. 09b6ff9 Disable PLC for iSAC by henrik.lundin@webrtc.org · 10 years ago
  50. ee0c5af Remove unused version.py script. by kjellander@webrtc.org · 10 years ago
  51. aa0bbab Fix build failure by jmarusic@webrtc.org · 10 years ago
  52. a4bef3e AcmReceiver: use std::map instead of an array to keep the list of decoders by jmarusic@webrtc.org · 10 years ago
  53. 3335a4f Prevent asserting on unset start bitrate. by pbos@webrtc.org · 10 years ago
  54. 50ed0d9 Roll chromium_revision 6311617..da9a1c0 (321517:321718) by kjellander@webrtc.org · 10 years ago
  55. e5e92bd Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) by kjellander@webrtc.org · 10 years ago
  56. cfde27e Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. by kjellander@webrtc.org · 10 years ago
  57. 38492c5 Re-land 8810 "- Add a SetPriority method to ThreadWr..." by tommi@webrtc.org · 10 years ago
  58. 90a1cb4 Revert 8810 "- Add a SetPriority method to ThreadWrapper" by tommi@webrtc.org · 10 years ago
  59. b789f62 Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." by tommi@webrtc.org · 10 years ago
  60. 0c34001 Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." by tommi@webrtc.org · 10 years ago
  61. 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 · 10 years ago
  62. 4553941 Document the 'int' return value of Resampler methods. by wtc@chromium.org · 10 years ago
  63. 3200a64 Minor fix for MIPS Android build. by andrew@webrtc.org · 10 years ago
  64. 4ddc938 Support VP8 hardware encoding and decoding on IA devices. by glaznev@webrtc.org · 10 years ago
  65. b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 10 years ago
  66. b6817d7 - Add a SetPriority method to ThreadWrapper by tommi@webrtc.org · 10 years ago
  67. 66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 10 years ago
  68. 8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  69. a3209a2 Release buffer pool in Vp8DecoderImpl::Release(). by pbos@webrtc.org · 10 years ago
  70. 8904290 Make screenshare target bitrate experiment always on by pbos@webrtc.org · 10 years ago
  71. d9c5024 Roll chromium_revision bd49b12..6311617 (320783:321517) by kjellander@webrtc.org · 10 years ago
  72. 9f9ea7e Clean up webrtc external capture. by perkj@webrtc.org · 10 years ago
  73. 443ad40 Remove FullStackTest frame pointer handles. by pbos@webrtc.org · 10 years ago
  74. 6231fb6 Prevent crashes when copying a zero-size frame. by pbos@webrtc.org · 10 years ago
  75. 6069032 Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  76. 4ab23d0 Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  77. bd8c865 Remove build-time beamformer flags. by andrew@webrtc.org · 10 years ago
  78. 04c5098 Add the Ooura FFT to RealFourier. by andrew@webrtc.org · 10 years ago
  79. ba86031 Whitespace change to trigger new Git pollers (2). by kjellander@webrtc.org · 10 years ago
  80. cf3fb9b Whitespace change to trigger new Git pollers. by kjellander@webrtc.org · 10 years ago
  81. 80d9aee Adds full-duplex unit test to AudioDeviceTest on Android by henrika@webrtc.org · 10 years ago
  82. 361981f Use scoped_ptr for ThreadWrapper::CreateThread. by tommi@webrtc.org · 10 years ago
  83. c7d5a73 Disable flaky test on DrMemory bots by tina.legrand@webrtc.org · 10 years ago
  84. 27c0be9 Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. by tommi@webrtc.org · 10 years ago
  85. 0c26299 Disabling two flaky tests in libjingle_media_unittest. by tina.legrand@webrtc.org · 10 years ago
  86. 17c64d1 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" by magjed@webrtc.org · 10 years ago
  87. c7157da Use atomic operations for setting/reading the trace filter. by tommi@webrtc.org · 10 years ago
  88. 9afaee7 Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() by jmarusic@webrtc.org · 10 years ago
  89. d21406d Remove command-line tool 'video_coding_test'. by pbos@webrtc.org · 10 years ago
  90. c4709a2 Split C++ class from macro overrides to fix Chromium build by tommi@webrtc.org · 10 years ago
  91. 5506a93 Expose ViECaptureImpl::DisconnectCaptureDevice() to JNI of WebRTCDemo and call it before releasing camera to deregister the corresponding framecallback. Also stop camera after stop remote rendering as the correct termination order. by braveyao@webrtc.org · 10 years ago
  92. 8cc47e9 Objective-C readability review. by tkchin@webrtc.org · 10 years ago
  93. 2a8a46d vp8: Add missing call to SetUsageMessage(). by kjellander@webrtc.org · 10 years ago
  94. 8f76cd2 Renaming neteq_opus_fec_quality_test. by minyue@webrtc.org · 10 years ago
  95. 840da7b Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  96. 143451d Base start bitrate on last observed bitrate. by pbos@webrtc.org · 10 years ago
  97. 5a477a0 DCHECK frame parameters instead of return codes. by pbos@webrtc.org · 10 years ago
  98. 4346d92 Use SendTimeHistory to keep track of send times in simulations. by stefan@webrtc.org · 10 years ago
  99. f189933 Removing henrik.lundin from OWNERS in video_coding/* by henrik.lundin@webrtc.org · 10 years ago
  100. af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago