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gerrit-public.fairphone.software / platform / external / webrtc / a7ee14ebf540e43862cefb2516f18f770230f198
  1. 090c940 Sort method declarations/definitions in VideoReceiveStream. by brandtr · 8 years ago
  2. 3373eaa Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) by ehmaldonado · 8 years ago
  3. 36cb55d GN: Refactor modules_unittests to eliminate package boundary violations. by ehmaldonado · 8 years ago
  4. 2676461 Moving build_aar.py to new location by mbonadei · 8 years ago
  5. bfb11b2 Call RtpStreamReceiver.AddReceiveCodec() with codec_params. by johan · 8 years ago
  6. d160fd7 Disabled EndToEndTest.ReceivesFlexfecAndSendsCorrespondingRtcp on Asan by aleloi · 8 years ago
  7. 8294952 Roll chromium_revision 66ac8d1a05..516550732e (445993:446004) by buildbot · 8 years ago
  8. e5dc3ce Fixing cross-compiling issues on android arm by mbonadei · 8 years ago
  9. c07cc56 Roll chromium_revision 34f2476fb5..66ac8d1a05 (445972:445993) by buildbot · 8 years ago
  10. 7d25426 Delete unneeded includes of base/common.h. by nisse · 8 years ago
  11. b63a8ac Moving gn_isolate_map.pyl to tools-webrtc/mb by Mirko Bonadei · 8 years ago
  12. 8d36274 Roll chromium_revision 900b07d425..34f2476fb5 (445935:445972) by buildbot · 8 years ago
  13. 630f46a Moving adb_shell script to tools-webrtc by mbonadei · 8 years ago
  14. 18e83ea Moving sanitizers from build/ to base/ by mbonadei · 8 years ago
  15. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  16. 6b75f4d Roll chromium_revision 34daea6e90..900b07d425 (445884:445935) by buildbot · 8 years ago
  17. 3a1eb04 Roll chromium_revision e003d59373..34daea6e90 (445829:445884) by buildbot · 8 years ago
  18. eaae505 Removing unused variable OUTPUT_LIB by VladimirTechMan · 8 years ago
  19. 0dabfbf Roll chromium_revision 15a29620cf..e003d59373 (445766:445829) by buildbot · 8 years ago
  20. ed111da Adding deadbeef@webrtc.org to webrtc/base/OWNERS. by deadbeef · 8 years ago
  21. eadcf36 Roll chromium_revision 057e94297e..15a29620cf (445739:445766) by buildbot · 8 years ago
  22. ae25512 Roll chromium_revision 319b885718..057e94297e (445689:445739) by buildbot · 8 years ago
  23. e9f36d5 Make sure min and max bitrate is always set for audio. by stefan · 8 years ago
  24. e256bc5 Delete left-over using declaration. by nisse · 8 years ago
  25. a388310 Added api/webrtcsdp.h forwarding header to work around upstream projects. by ossu · 8 years ago
  26. 9aa3f0a Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 8 years ago
  27. b54c63f Moving no_op_function.cc out of webrtc/build by mbonadei · 8 years ago
  28. dabbea6 Moving whitespace file up by one folder by mbonadei · 8 years ago
  29. 69dc7db Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 8 years ago
  30. e6b4723 Statically linked zxing. Without dependency on libMagick by mandermo · 8 years ago
  31. 4b7c952 Reland of "Log audio network adapter decisions in event log." by minyue · 8 years ago
  32. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  33. 62d02c3 Unit test out of band H264 SPS,PPS within RtpStreamReceiver. by johan · 8 years ago
  34. 822d258 Move webrtc/build/android -> tools-webrtc/android by mbonadei · 8 years ago
  35. 81eab61 Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo. by brandtr · 8 years ago
  36. 0608ffd Roll chromium_revision 59592eaa98..319b885718 (445345:445689) by buildbot · 8 years ago
  37. 365aebd Make CongestionController::remote_bitrate_estimator_ a non-pointer. by nisse · 8 years ago
  38. d2b092f Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. by johan · 8 years ago
  39. 15389c0 Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. by nisse · 8 years ago
  40. 568c9e7 New simulators to test BWE at low bitrates (15-50kbps range). by terelius · 8 years ago
  41. a4a7538 Android: Script for building libwebrtc.aar. by sakal · 8 years ago
  42. e04064d Revert of Delete unused class/template ScopedMessageData. (patchset #1 id:1 of https://codereview.webrtc.org/2652663002/ ) by aleloi · 8 years ago
  43. dc2b3f3 Delete unused class CompositeMediaEngineWithFakeVoiceEngine. by nisse · 8 years ago
  44. d83fb92 Delete unused class/template ScopedMessageData. by nisse · 8 years ago
  45. c23b0b2 Delete unused classes DesktopId and ScreencastEventCatcher. by nisse · 8 years ago
  46. ad45228 Moving get_landmines.py (build/ -> tools-webrtc/) by mbonadei · 8 years ago
  47. 2b75526 Add linux_memcheck as default trybot. by Henrik Kjellander · 8 years ago
  48. 914d49d Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ ) by kjellander · 8 years ago
  49. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  50. 4c78702 iOS: Add MedianSlopeFilter field trial. by tkchin · 8 years ago
  51. 5c4f24a Move implmentation specific constants out of rtp_header_extension.h by danilchap · 8 years ago
  52. f53d737 H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. by johan · 8 years ago
  53. e1405ad Removed double-special-casing of ISAC in libjingle and WebRtcVoE. by ossu · 8 years ago
  54. cb893ee Removing unused code from webrtc/build by mbonadei · 8 years ago
  55. 1bed2e4 video_loopback: fall back to fake capturer if we can't open camera. by sprang · 8 years ago
  56. 435ddf9 Add TransportFeedbackPacketLossTracker. by minyue · 8 years ago
  57. ed582f7 Script to start stubbed loopback video test with Espresso by mandermo · 8 years ago
  58. 0ebdf27 Delete or update left-over ASSERT use and comments. by nisse · 8 years ago
  59. da25006 Fixed public_deps for libjingle_peerconnection{,_api} by ossu · 8 years ago
  60. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
  61. 9c3d4c4 Stop leaking FlexfecReceiveStream objects after call shutdown. by brandtr · 8 years ago
  62. a067013 Minor style change suggested by internal static analysis tool. by aleloi · 8 years ago
  63. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  64. f49ff26 GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false by ehmaldonado · 8 years ago
  65. fd870db Add metric for decode time and max decode time in video quality tests. by philipel · 8 years ago
  66. 0112403 Minor style change suggested by internal static analysis tool. by aleloi · 8 years ago
  67. a31cdbc Roll chromium_revision dcc5978539..59592eaa98 (445328:445345) by buildbot · 8 years ago
  68. 0b56279 Catch failure to load native dependencies. by sakal · 8 years ago
  69. de8ca92 New script to count usage of C++ classes. by nisse · 8 years ago
  70. b55bd97 Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ ) by mbonadei · 8 years ago
  71. 5d0f2e8 Roll chromium_revision 269b6bc66e..dcc5978539 (445317:445328) by buildbot · 8 years ago
  72. c152434 Roll chromium_revision 7649e76842..269b6bc66e (445027:445317) by buildbot · 8 years ago
  73. 3e4faae Fixing memory leak in FakeTransportController. by deadbeef · 8 years ago
  74. 8662f94 Only set certificate on DTLS transport if fingerprint is found in SDP. by deadbeef · 8 years ago
  75. 2197e91 Remove dead code for GtkVideoRenderer. by pbos · 8 years ago
  76. f33491e Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ ) by deadbeef · 8 years ago
  77. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  78. cd3180c PATENTS: fix reference by philipp.hancke · 8 years ago
  79. 7bcdb69 Ignore ufrag/password in "a=candidate" lines in SDP. by deadbeef · 8 years ago
  80. 0fc04b7 Finalize the support for building WebRTC library for iOS with bitcode by VladimirTechMan · 8 years ago
  81. f64941f RTCMediaStreamTrackStats.framesDecoded collected. by hbos · 8 years ago
  82. aea1a01 Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS by magjed · 8 years ago
  83. 3c9151b Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ ) by mbonadei · 8 years ago
  84. a62a82b Creating libwebrtc bundle jar by mbonadei · 8 years ago
  85. fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 8 years ago
  86. 2d4d653 Fix msan flake in rtcstats_integrationtest.cc. by hbos · 8 years ago
  87. c854ac3 Stop camera onStop instead of onPause. by sakal · 8 years ago
  88. 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 8 years ago
  89. 7319f26 Roll chromium_revision 780d18a4ff..7649e76842 (445004:445027) by buildbot · 8 years ago
  90. 30fe5e0 Prevent downstream linter warnings. by sakal · 8 years ago
  91. 3556406 Camera1Session: Fix camera sometimes getting stopped twice. by sakal · 8 years ago
  92. 9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 8 years ago
  93. fd6c94d Allow more config changes for CallActivity. by sakal · 8 years ago
  94. 3e92290 Load library dependencies in AppRTCMobile. by sakal · 8 years ago
  95. be850e1 Clear out cached codecs when calculating new codec lists. by noahric · 8 years ago
  96. 204030a Roll chromium_revision bdeae63b37..780d18a4ff (444971:445004) by buildbot · 8 years ago
  97. a01d2f5 Roll chromium_revision 34215edf2e..bdeae63b37 (444898:444971) by buildbot · 8 years ago
  98. 888874f Allow GCC 4.9 to compile Chromium by floppymaster · 8 years ago
  99. 8944ab3 Roll chromium_revision 1a7fcf6220..34215edf2e (444851:444898) by buildbot · 8 years ago
  100. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago