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gerrit-public.fairphone.software / platform / external / webrtc / d7a75d798b9434c441970f996771e625280df24d
  1. d7a75d7 Roll chromium_revision c6ec25c..da1acd5 (371549:371832) by kjellander · 10 years ago
  2. 7b3c72f Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #4 id:60001 of https://codereview.webrtc.org/1581693006/ ) by deadbeef · 10 years ago
  3. 42265a8 Adding "first packet received" notification to PeerConnectionObserver. by Taylor Brandstetter · 10 years ago
  4. 80f1db9 Include relay protocol type when computing the turn candidate foundation. by Honghai Zhang · 10 years ago
  5. 3afc8c4 Consolidate SetSendParameters into one setter. by Peter Boström · 10 years ago
  6. ec2922f Change PeerConnectionFactory.setVideoHwAccelerationOptions to create shared Egl context for harware encoders and decoders. by Per · 10 years ago
  7. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 10 years ago
  8. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 10 years ago
  9. f5dca48 Add transport sequence number on the non-pacer path of the rtp sender. by Stefan Holmer · 10 years ago
  10. 1c39098 Use rtc::time for all your timing needs! by Erik Språng · 10 years ago
  11. d673b0f [rtp_rtcp] Fix potentional time difference between rtp and rtcp packets. by Danil Chapovalov · 10 years ago
  12. b11e97a Move talk/media/webrtc/OWNERS to talk/media. by Peter Boström · 10 years ago
  13. 0b518bf Remove incorrect cast to AsyncSocketAdapter. by Peter Boström · 10 years ago
  14. bab934b H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. by hbos · 10 years ago
  15. fab0a28 Fix BasicNetworkManager not to spam logs when internet is unreacheable. by Sergey Ulanov · 10 years ago
  16. 3ea1852 Sync build_ios_libs.sh script with http://webrtc.org/native-code/ios/ by hjon · 10 years ago
  17. 4cb3e39 Fix compilation if HAVE_WEBRTC_VIDEO is not defined. by jbauch · 10 years ago
  18. 6d49a8e Update API for Objective-C RTCConfiguration. by hjon · 10 years ago
  19. 7b582a2 Roll chromium_revision 2ca77c1..c6ec25c (371488:371549) by kjellander · 10 years ago
  20. a2c5523 Allow packets to be reordered in the fake network pipe. by philipel · 10 years ago
  21. 7fd8817 Fix type of local encoded length variable from uint32_t to size_t. by asapersson · 10 years ago
  22. 59b2d3e Remove zero-divide in VCMContentMetricsProcessing. by Peter Boström · 10 years ago
  23. 8327713 AudioCodingModuleImpl: Put CodecManager and Rent-A-Codec in a separate struct by kwiberg · 10 years ago
  24. d0c7bba [rtp_rtcp] Dlrr::SubBlock struct renamed to ReceiveTimeInfo by Danil Chapovalov · 10 years ago
  25. 5c7f110 Roll chromium_revision fb2e77c..2ca77c1 (371273:371488) by kjellander · 10 years ago
  26. 6a07f12 AudioCodingModuleImpl: Initialize encoder_stack_ to nullptr by kwiberg · 10 years ago
  27. 2bdcfad Revert of Removing webrtc::AudioFrame::energy_. (patchset #2 id:20001 of https://codereview.webrtc.org/1589953002/ ) by terelius · 10 years ago
  28. ffa3fdc Reallocate encoded buffer size if needed for VP8. Initially set to the input image size. by asapersson · 10 years ago
  29. e791ffd Remove non-monotonic clock support by sprang · 10 years ago
  30. 4fd6cda Add tracing to VCMGenericEncoder::Release. by Peter Boström · 10 years ago
  31. 86956de Small cleanup in VP9EncoderImpl::GetEncodedLayerFrame. by asapersson · 10 years ago
  32. bacae81 Remove webrtc::AudioFrame::energy_. by minyue · 10 years ago
  33. 58a80b5 Roll chromium_revision 717238e..fb2e77c (370438:371273) by kjellander · 10 years ago
  34. 85b22e2 Remove vp8_factory.{cc,h}. by Peter Boström · 10 years ago
  35. b332e5d Roll chromium_revision 6a04368..717238e (370362:370438) + tcmalloc by primiano · 10 years ago
  36. 28ba927 Switch to use new implementation in metrics.h. by asapersson · 10 years ago
  37. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 10 years ago
  38. 7d0d0e0 Remove dead code from webrtc/base/timing.* by tommi · 10 years ago
  39. 9de632a Deleted unused enums MediaChannelOptions and VoiceMediaChannelOptions, by nisse · 10 years ago
  40. 7a83951 Fix a bug in webrtc::ByteReader by henrik.lundin · 10 years ago
  41. f91e6d0 Enable cpplint for webrtc/modules/bitrate_controller and fix all uncovered cpplint errors. by jbauch · 10 years ago
  42. e373dc2 Update API for Objective-C RTCDataChannel. by hjon · 10 years ago
  43. 38b39d5 Temporary hack to avoid assert errors when time moves backwards. by sprang · 10 years ago
  44. cc71c41 Revert "Disable P2PTransport...TestFailoverControlledSide on Memcheck" by tnakamura · 10 years ago
  45. 0a37497 Deleted unused method SetDumpPath and unneeded includes. by nisse · 10 years ago
  46. c8930ba Disable WebRtcSessionTest.TestStunError on Win. by minyue · 10 years ago
  47. 9846845 Calculate audio levels in AEC in time domain. by minyue · 10 years ago
  48. 5447934 Remove implementation of CriticalSectionWrapper and use rtc::CriticalSection by tommi · 10 years ago
  49. 7406b96 CriticalSection: Use types+methods from base/platform_thread*.*. by tommi · 10 years ago
  50. 32e590e class doesn't rely on structures in RTCPUtility to store data. by Danil Chapovalov · 10 years ago
  51. 3fe2c6a VideoProcessorImpl using EncodedImage::GetBufferPaddingBytes. by hbos · 10 years ago
  52. ed281e9 New lock implementation for mac. by tommi · 10 years ago
  53. 2bf9a5f Update API for Objective-C RTCMediaStream. by Jon Hjelle · 10 years ago
  54. ca91e38 Update API for Objective-C RTCAudioTrack and RTCVideoTrack. by Jon Hjelle · 10 years ago
  55. 97888bd Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video. by Tommi · 10 years ago
  56. 7ac8bab Move RTCAVFoundationCapturer to webrtc/api/objc. by Jon Hjelle · 10 years ago
  57. 891a446 Update/move RTCVideoRendererAdapter to webrtc/api/objc. by Jon Hjelle · 10 years ago
  58. 31fc21f Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ by tommi · 10 years ago
  59. 8947a01 Fixing an uninitialized variable in webrtcsession_unittest. by deadbeef · 10 years ago
  60. fa15669 Fix probing breakage with send-side BWE introduced by r11322. by stefan · 10 years ago
  61. fea3dd8 Fix a bug in InputAudioFile::Read by henrik.lundin · 10 years ago
  62. af9e663 Make rtc::CriticalSection lockable from f() const. by Peter Boström · 10 years ago
  63. 3c16978 Remove cast to LocalAudioSource from AudioRtpSender. by Tommi · 10 years ago
  64. 32be07b Remove RentACodec::GetEncoderStack by kwiberg · 10 years ago
  65. 693a114 Add stefan@webrtc.org to webrtc/test/OWNERS. by Peter Boström · 10 years ago
  66. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 10 years ago
  67. d664836 Added EncodedImage::GetBufferPaddingBytes. by hbos · 10 years ago
  68. 429c345 Fixes a bug which incorrectly logs incoming RTCP as outgoing. by terelius · 10 years ago
  69. b304e26 Roll chromium_revision 1728ddf..4623ce8 (370595:370665) by kjellander · 10 years ago
  70. 1f150b3 Add new NetEq resources to modules_unittests.isolate. by kjellander · 10 years ago
  71. 902c03e rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni. by hbos · 10 years ago
  72. 0b98cf7 Delete CaptureRenderAdapter::VideoRenderInfo struct, it is unused since the recent deletion of SetSize. by nisse · 10 years ago
  73. 5082c83 Make type and constructors in EglBase14 public. by noahric · 10 years ago
  74. becf9ee Roll chromium_revision 6a04368..1728ddf (370362:370595) by kjellander · 10 years ago
  75. d9f641e Reallocate encoded buffer size if needed. Initially set to the input image size. by asapersson · 10 years ago
  76. d26fadb Delete GetRenderer method, used only by the tests. by nisse · 10 years ago
  77. 057ecf0 Making WebRtcSession fire a destroyed signal. by deadbeef · 10 years ago
  78. da99da8 Update API for Objective-C RTCPeerConnectionFactory. by Jon Hjelle · 10 years ago
  79. 065aacc Move RTCVideoSource to webrtc/api/objc. by Jon Hjelle · 10 years ago
  80. d8dccd5 uses standard types instead of RTCPUtility type to store data. by danilchap · 10 years ago
  81. 72c08ed Reenables several NetEq unittests on android. by ivoc · 10 years ago
  82. 32f8154 Support REMB in combination with send-side BWE. by stefan · 10 years ago
  83. a5dec16 Name SimulcastEncoderApdater on InitEncode. by Peter Boström · 10 years ago
  84. a2b4c40 Roll chromium_revision 15d94b7..6a04368 (370289:370362) by kjellander · 10 years ago
  85. 9090e0b Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding. by Tommi · 10 years ago
  86. 84df580 Switch to rtc::CriticalSection in IncomingVideoStream and remove one lock. by tommi · 10 years ago
  87. e849332 Remove ConditionVariableWrapper. by Tommi · 10 years ago
  88. 63cb434 Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/ by tommi · 10 years ago
  89. 1d61a51 Send key frame if time difference between incoming frames exceeds a certain limit. by asapersson · 10 years ago
  90. 436ff31 Update exclude files for renamed test by kjellander · 10 years ago
  91. a927dcf Roll chromium_revision 542b77a..15d94b7 (370158:370289) by kjellander · 10 years ago
  92. f0b8a37 Allow disabling denoiser when it is enabled. by jackychen · 10 years ago
  93. 3a6bf2d Enable full screen windows to be shown in window picker for mac. Before this patch a full screen window can be shared if sharing is started before the window is entered into full screen mode, but not if it's already in full screen. by niklas.enbom · 10 years ago
  94. 95c8b40 Roll chromium_revision f527e86..542b77a (370073:370158) by kjellander · 10 years ago
  95. f01ea4f Remove use of ConditionVariableWrapper and CriticalSectionWrapper from UdpSocket2Windows. by Tommi · 10 years ago
  96. cd255cc Remove unused ConditionVariableWrapper on POSIX platforms by tommi · 10 years ago
  97. 7b971e7 Remove extra_options from VideoCodec. by Peter Boström · 10 years ago
  98. ee5a309 Make CriticalSectionWrapper non-virtual. by Tommi · 10 years ago
  99. dd45eb6 Remove use-after-free when quality tests stall. by Peter Boström · 10 years ago
  100. 8a2c31d Make it possible to run peerconnection_unittests on Android. by perkj · 10 years ago