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gerrit-public.fairphone.software / platform / external / webrtc / 55e9a7dc4ba922599d4fbcf25f13629ba210fc64
  1. 55e9a7d Add Android VideoRendererGui events. by Alex Glaznev · 9 years ago
  2. d332580 Add stats overlay to iOS AppRTCDemo. by Zeke Chin · 9 years ago
  3. 60d9b33 Integrate Intelligibility with APM by ekmeyerson · 9 years ago
  4. 03bb7c7 Add LoudestFilter in ConferenceTransport by minyue · 9 years ago
  5. 4c530dc Delete dummy dtlsidentityservice.[cc,h] files. by hbos · 9 years ago
  6. d5031fc Android VideoRendererGui: Add dispose function by magjed · 9 years ago
  7. af5c035 VideoCapturerAndroid: Release queued camera frames when stopCapture() is called by magjed · 9 years ago
  8. 38f8893 WebRTC Bug 4865 by Guo-wei Shieh · 9 years ago
  9. ee8c6d3 In PeerConnectionTestWrapper, put audio input on a separate thread. by deadbeef · 9 years ago
  10. 7437588 Adding locking to webrtc::voe::Channel to fix race conditions by deadbeef · 9 years ago
  11. c558af8 Removing DtlsIdentityService[Interface] which has been replaced by DtlsIdentityStore[Interface/Impl]. by hbos · 9 years ago
  12. cf7f54d Use RtcpPacket to send RPSI in RtcpSender by sprang · 9 years ago
  13. e2a8be1 Revert of AppRTCDemo: Render each video in a separate SurfaceView (patchset #4 id:120001 of https://codereview.webrtc.org/1257043004/ ) by magjed · 9 years ago
  14. d941b76 Fix distortions of remote stream with odd size dimensions by budnyjj · 9 years ago
  15. 8a2cd3d Revert H.264 HW encoder setting to CBR mode. by Alex Glaznev · 9 years ago
  16. d6b243f Enabling screensharing perf test. by ivica · 9 years ago
  17. 05bfbe4 AppRTCDemo: Render each video in a separate SurfaceView by magjed · 9 years ago
  18. fa30180 Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by pthatcher · 9 years ago
  19. cc4ebad Empty dtlsidentityservice.h/cc files added, to be removed once chromium gyp files don't reference it. by Henrik Boström · 9 years ago
  20. 5e56c59 DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface). by Henrik Boström · 9 years ago
  21. 0365a27 Use RtcpPacket to send SLI in RtcpSender by sprang · 9 years ago
  22. 4bc66fc Fix data race in AMP. by Michael Graczyk · 9 years ago
  23. 4de6622 Fix a bug in computing audio delay on ios device. Converts seconds to by Jiawei Ou · 9 years ago
  24. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  25. 4cee419 Separating voice activity flag from audio level in RtpHeaderExtension. by Minyue · 9 years ago
  26. c2ee2c8 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well. by Peter Thatcher · 9 years ago
  27. eb04d68 Moved project configs to infra/config branch by nodir · 9 years ago
  28. 25c96d0 Add thread checker to StatsCollection. by jbauch · 9 years ago
  29. 2328a94 Add average rtt to CallStatsObserver and an average rtt histogram. by stefan · 9 years ago
  30. 0482dcc Enable HW H.264 decoding on Intel platforms. by Alex Glaznev · 9 years ago
  31. 8381b37 Removed bjornv from OWNERS and added two new owners by peah · 9 years ago
  32. 2e1d8bb Suppress a race in libjingle_peerconnection_unittest by henrik.lundin · 9 years ago
  33. fcf8ece AndroidVideoCapturer: Return frames that have been dropped by magjed · 9 years ago
  34. c937139 Regenerate bind.h using pump.py BUG=webrtc:4690 R=pthatcher@webrtc.org by Fredrik Solenberg · 9 years ago
  35. a873644 Move all the examples from the talk directory into the webrtc examples directory. by Donald E Curtis · 9 years ago
  36. 5b4ce33 DtlsIdentityStoreInterface added. by Henrik Boström · 9 years ago
  37. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 9 years ago
  38. bd10ee8 Tiny cleanups. by Fredrik Solenberg · 9 years ago
  39. 62dae19 Use RtcpPacket to send FIR in RtcpSender by sprang · 9 years ago
  40. ef7228c Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better. by sprang · 9 years ago
  41. 907dcfd Increase packet limit in jitter buffer. by sprang · 9 years ago
  42. 37ec733 VideoCapturerAndroid: Check if data is null in onPreviewFrame() by magjed · 9 years ago
  43. 0c85020 Add list of devices with HW H.264 encoder non suitable for WebRTC. by Alex Glaznev · 9 years ago
  44. 8d62971 Fix race condition in EndToEndTest.AssignsTransportSequenceNumbers by Erik Språng · 9 years ago
  45. b19eba3 Fix Turn TCP port issue. by honghaiz · 9 years ago
  46. 867fb52 Add support for transport wide sequence numbers by sprang · 9 years ago
  47. d67a219 Switch to base/logging.h in neteq_impl.cc by Henrik Lundin · 9 years ago
  48. 62cde2c Disabling VP9 perf test by ivica · 9 years ago
  49. 503726c Fix the generation mismatch assertion error. by honghaiz · 9 years ago
  50. 72aa9a6 Use RtcpPacket to send PLI in RtcpSender by Erik Språng · 9 years ago
  51. a9455ab Integration of VP9 packetization. by asapersson · 9 years ago
  52. 2386a45 Supporting Pause/Resume, Sending Estimate logging. Corrected plot colors by Cesar Magalhaes · 9 years ago
  53. a12ba55 Added protection for GetCapabilities() failure. by dkirovbroadsoft · 9 years ago
  54. 5f5f11c FEC protect H264 delta frames as well. by pbos · 9 years ago
  55. 3641185 Includes webrtc/build/protoc.gypi instead of build/protoc.gypi by Bjorn Terelius · 9 years ago
  56. b933667 Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly." by Bjorn Terelius · 9 years ago
  57. 9a6e741 Move audio_coding_module.gypi from main/acm2 to main/. by Peter Boström · 9 years ago
  58. e2cb1f1 Efficient Metric Recorder by Cesar Magalhaes · 9 years ago
  59. 028cf48 Added FullStack performance test for screensharing with VP9 by ivica · 9 years ago
  60. c159b04 Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly. by Bjorn Terelius · 9 years ago
  61. ee66016 Added IsInBeam to mock_nonlinear_beamformer.h by bloch · 9 years ago
  62. d635895 Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests. by sprang · 9 years ago
  63. 49c0ce3 Revert "Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests." by Erik Språng · 9 years ago
  64. 8993413 Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests. by Erik Språng · 9 years ago
  65. a3b8769 Add packetization and coding/decoding of feedback message format. by Erik Språng · 9 years ago
  66. f1828e8 Prevent OOB reads for truncated H264 STAP-A packets. by pbos · 9 years ago
  67. f38ea3c Add support for VP9 packetization/depacketization. by asapersson · 9 years ago
  68. 95b8718 Fix to "Removing AudioMixerStatusReceiver and ParticipantStatistics" by Minyue Li · 9 years ago
  69. 4540ffa Removing AudioMixerStatusReceiver and ParticipantStatistics. by Minyue · 9 years ago
  70. d40af69 Split MoveReadPosition into Forward and Backward versions. by andrew · 9 years ago
  71. b3cc77f Re-enable WebRtcIsacfix_AllpassFilter2FixDec16Neon by Zhongwei Yao · 9 years ago
  72. a446609 When we trace to file, add eol of each trace message. by Brave Yao · 9 years ago
  73. b3b79b6 Clean up the Config to enable 48kHz support in AudioProcessing by aluebs · 9 years ago
  74. ef35f06 Remove webrtc::Config from ViEChannelGroup. by pbos · 9 years ago
  75. 081af25 Remove kProtectionKey* and VCMKeyRequestMode. by pbos · 9 years ago
  76. fa37e33 Add pbos@webrtc.org to webrtc/video_engine/OWNERS. by pbos · 9 years ago
  77. fe0c905 Improve probing by ignoring small packets which otherwise break the mechanism. by stefan · 9 years ago
  78. b28678c Add unittest to GlRectDrawer by magjed · 9 years ago
  79. 013a580 VideoCapturerAndroid: Revert elapsedRealtimeNanos to elapsedRealtime by magjed · 9 years ago
  80. d55ce2d BWE Simulation Framework: Standard plot logging by Cesar Magalhaes · 9 years ago
  81. 7a1c24f Remove "multichannel" from parameter to match interface name. by andrew · 9 years ago
  82. e2b34b7 Bug fix: camera frames are dropped before wideo encoder. by jackychen · 9 years ago
  83. 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 9 years ago
  84. cfd5f96 Ignore packets with reordered timestamps when doing BWE. by stefan · 9 years ago
  85. a38233a Removed extended jitter report from RtcpSender. by Erik Språng · 9 years ago
  86. 6718e97 Add encode and decode time to histograms stats: by asapersson · 9 years ago
  87. c3f46a9 iOS: Move AppRTC logging methods to public headers. by tkchin · 9 years ago
  88. 28bae02 Remove CircularFileStream / replace it with CallSessionFileRotatingStream. by tkchin · 9 years ago
  89. 3ab2f14 Remove C++11 calls from intelligibility_utils by ekmeyerson · 9 years ago
  90. 86c6d33 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 9 years ago
  91. fcfdb08 Update AUTHORS file. by tkchin · 9 years ago
  92. d6fc47e Remove base channel for video receivers. by pbos · 9 years ago
  93. 59adf34 Evaluation test cases. by Cesar Magalhaes · 9 years ago
  94. 66f438f Revert of Fixing scenario where track is rejected and later un-rejected. (patchset #5 id:80001 of https://codereview.webrtc.org/1231613002/) by magjed · 9 years ago
  95. 64e753c Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/) by magjed · 9 years ago
  96. b21fd94 Temporarily disable ScreenshareSlides on Android. by Peter Boström · 9 years ago
  97. c204754 Allow more than 2 input channels in AudioProcessing. by Michael Graczyk · 9 years ago
  98. 0b6a204 Configure AudioProcessing directly in agc_harness. by andrew · 9 years ago
  99. b297c5a Miscellaneous changes split from https://codereview.webrtc.org/1230503003 . by pkasting · 9 years ago
  100. 7c5304c Allow webrtc compilation with stlport by Jared Duke · 9 years ago