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gerrit-public.fairphone.software / platform / external / webrtc / e702b30fec47933114161691369bd27a90371862
  1. 66d46ae Roll chromium_revision e3bc84e363..b2f66c7a95 (446911:446919) by buildbot · 8 years ago
  2. b409e23 Roll chromium_revision d1351ea096..e3bc84e363 (446900:446911) by buildbot · 8 years ago
  3. 286299d Roll chromium_revision 647709aaba..d1351ea096 (446860:446900) by buildbot · 8 years ago
  4. 4460e7f Roll chromium_revision 14ab9f9226..647709aaba (446784:446860) by buildbot · 8 years ago
  5. 7d4a327 Roll chromium_revision 22ab374ddc..14ab9f9226 (446723:446784) by buildbot · 8 years ago
  6. 0c1d060 Enable Android H264 High profile decoder by glaznev · 8 years ago
  7. 62a5dd2 Roll chromium_revision 9ff019ad14..22ab374ddc (446676:446723) by buildbot · 8 years ago
  8. 869c30f Roll chromium_revision 087876b708..9ff019ad14 (446653:446676) by buildbot · 8 years ago
  9. 16b0221 Prioritize video packets when sending padding or preemptive retransmits. by stefan · 8 years ago
  10. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 8 years ago
  11. 5a2c506 Set the start bitrate to the delay-based BWE. by stefan · 8 years ago
  12. b0ae920 RTCRTPStreamStats.mediaTrackId renamed to trackId. by hbos · 8 years ago
  13. 55d1ebb Enable periodic bitrate probing when application limited for audio BWE. by stefan · 8 years ago
  14. 206b0d7 Roll chromium_revision 6f11aa45a2..087876b708 (446630:446653) by buildbot · 8 years ago
  15. b621c3f Move Android tests under sdk/android. by sakal · 8 years ago
  16. 1474212 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) by brandtr · 8 years ago
  17. 986012d iOS AppRTCMobile: Enable H264 High profile by magjed · 8 years ago
  18. 89da160 Disable flaky test VideoSendStreamTest.RemoveOverheadFromBandwidth. by aleloi · 8 years ago
  19. 69221db Adding second layer of the echo canceller 3 functionality. by peah · 8 years ago
  20. 270048c Roll chromium_revision a429302c3d..6f11aa45a2 (446615:446630) by buildbot · 8 years ago
  21. f00497c Improve bitrate probing for the audio-only case. by stefan · 8 years ago
  22. 27378f3 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ ) by philipel · 8 years ago
  23. 54a05b2 Add temporary AddRtxInfo member function to VideoReceiveStream::Config. by brandtr · 8 years ago
  24. 8351d4a Removing unused script by mbonadei · 8 years ago
  25. 5232fac Roll chromium_revision 15da4792dd..a429302c3d (446579:446615) by buildbot · 8 years ago
  26. 284acf6 Disable VideoSendStreamTest.RemoveOverheadFromBandwidth for memcheck by Henrik Kjellander · 8 years ago
  27. 2c08475 Roll chromium_revision 6e45e58670..15da4792dd (446506:446579) by buildbot · 8 years ago
  28. 6f9871f Roll chromium_revision 063a0ce5f9..6e45e58670 (446467:446506) by buildbot · 8 years ago
  29. efa5d48 Roll chromium_revision de229b494a..063a0ce5f9 (446382:446467) by buildbot · 8 years ago
  30. 8225c40 Remove svc_context.h include by johannkoenig · 8 years ago
  31. e497495 Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) by kjellander · 8 years ago
  32. 71ecaad Roll chromium_revision 59c132dd68..de229b494a (446339:446382) by buildbot · 8 years ago
  33. 192132e Fix for video protection_bitrate in BWE with overhead. by michaelt · 8 years ago
  34. 04bd836 Roll chromium_revision 4f907099f6..59c132dd68 (446310:446339) by buildbot · 8 years ago
  35. 736871c No op changelist test by Mirko Bonadei · 8 years ago
  36. fe2bef3 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
  37. 4703741 Minor style fix to please internal style tool. by aleloi · 8 years ago
  38. c3c8236 Replace src/ios copy with DEPS mirror. by Henrik Kjellander · 8 years ago
  39. 429600d Reland of Add experimental simulcast screen content mode by sprang · 8 years ago
  40. 4b15d9a Roll chromium_revision c755988946..4f907099f6 (446293:446310) by buildbot · 8 years ago
  41. b0dd784 Fixing path to be compliant with new location by Mirko Bonadei · 8 years ago
  42. 3e00528 Disable flaky test TestFrameBuffer2.OneUnorderedSuperFrame. by aleloi · 8 years ago
  43. ad524d6 Moving build/ios to tools-webrtc/ios by Mirko Bonadei · 8 years ago
  44. 7b58960 replay: output rtp header elements for errors by philipp.hancke · 8 years ago
  45. 09d6ef0 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ ) by philipel · 8 years ago
  46. ac9c211 Roll chromium_revision 785a834775..c755988946 (446260:446293) by buildbot · 8 years ago
  47. c3dfff3 Avoid multiple calls to webrtc::field_trial::FindFullName in RTPSender; it's inefficient to perform string comparison whenever we send a packet. by elad.alon · 8 years ago
  48. 52cdd3b Make video_quality_measurement buildable again. by brandtr · 8 years ago
  49. f3d622d Revert of Disabled two iOS tests due to bot breakage. Affected tests are (patchset #1 id:1 of https://codereview.webrtc.org/2652423002/ ) by aleloi · 8 years ago
  50. 327c450 Disabled EndToEndTest.{ReceivesFlexfec, ReceivesFlexfecAndSendsCorrespondingRtcp, CanReceiveUlpfec} due to breakages across several platforms. by aleloi · 8 years ago
  51. d3adbfb AppRTCMobile links against framework target. by kthelgason · 8 years ago
  52. 8e775e1 Disabled two iOS tests due to bot breakage. Affected tests are by aleloi · 8 years ago
  53. 4c4fdeb Infra CQ config: remove deprecated target_ref. by tandrii · 8 years ago
  54. bf73b7b Remove unused constants from video engine tests. by kthelgason · 8 years ago
  55. 535dbd3 add ImplementationName for VideoEncoderSoftwareFallbackWrapper. by kthelgason · 8 years ago
  56. 6895f36 Roll chromium_revision d61348de4e..785a834775 (446219:446260) by buildbot · 8 years ago
  57. acbd07f Roll chromium_revision a089b8668e..d61348de4e (446176:446219) by buildbot · 8 years ago
  58. 700dbea Roll chromium_revision 4afe10c428..a089b8668e (446129:446176) by buildbot · 8 years ago
  59. cc7213e Remove "video_capture" from modules' public_deps. by jianjun.zhu · 8 years ago
  60. 46fa413 Roll chromium_revision 495d2cc61a..4afe10c428 (446030:446129) by buildbot · 8 years ago
  61. 1a7f01c Validate the build type argument value in function build_webrtc by VladimirTechMan · 8 years ago
  62. 2d8cd58 Minor updates to the help message printed by build_ios_libs.sh (with -h) by VladimirTechMan · 8 years ago
  63. 148e370 iOS: Add SendSideBweWithOverhead field trial key by haysc · 8 years ago
  64. 329ec94 Roll chromium_revision 516550732e..495d2cc61a (446004:446030) by buildbot · 8 years ago
  65. e075430 Don't update the jitter estimate with frames containing retransmitted packets. by philipel · 8 years ago
  66. 090c940 Sort method declarations/definitions in VideoReceiveStream. by brandtr · 8 years ago
  67. 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
  68. 36cb55d GN: Refactor modules_unittests to eliminate package boundary violations. by ehmaldonado · 8 years ago
  69. 2676461 Moving build_aar.py to new location by mbonadei · 8 years ago
  70. bfb11b2 Call RtpStreamReceiver.AddReceiveCodec() with codec_params. by johan · 8 years ago
  71. d160fd7 Disabled EndToEndTest.ReceivesFlexfecAndSendsCorrespondingRtcp on Asan by aleloi · 8 years ago
  72. 8294952 Roll chromium_revision 66ac8d1a05..516550732e (445993:446004) by buildbot · 8 years ago
  73. e5dc3ce Fixing cross-compiling issues on android arm by mbonadei · 8 years ago
  74. c07cc56 Roll chromium_revision 34f2476fb5..66ac8d1a05 (445972:445993) by buildbot · 8 years ago
  75. 7d25426 Delete unneeded includes of base/common.h. by nisse · 8 years ago
  76. b63a8ac Moving gn_isolate_map.pyl to tools-webrtc/mb by Mirko Bonadei · 8 years ago
  77. 8d36274 Roll chromium_revision 900b07d425..34f2476fb5 (445935:445972) by buildbot · 8 years ago
  78. 630f46a Moving adb_shell script to tools-webrtc by mbonadei · 8 years ago
  79. 18e83ea Moving sanitizers from build/ to base/ by mbonadei · 8 years ago
  80. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  81. 6b75f4d Roll chromium_revision 34daea6e90..900b07d425 (445884:445935) by buildbot · 8 years ago
  82. 3a1eb04 Roll chromium_revision e003d59373..34daea6e90 (445829:445884) by buildbot · 8 years ago
  83. eaae505 Removing unused variable OUTPUT_LIB by VladimirTechMan · 8 years ago
  84. 0dabfbf Roll chromium_revision 15a29620cf..e003d59373 (445766:445829) by buildbot · 8 years ago
  85. ed111da Adding deadbeef@webrtc.org to webrtc/base/OWNERS. by deadbeef · 8 years ago
  86. eadcf36 Roll chromium_revision 057e94297e..15a29620cf (445739:445766) by buildbot · 8 years ago
  87. ae25512 Roll chromium_revision 319b885718..057e94297e (445689:445739) by buildbot · 8 years ago
  88. e9f36d5 Make sure min and max bitrate is always set for audio. by stefan · 8 years ago
  89. e256bc5 Delete left-over using declaration. by nisse · 8 years ago
  90. a388310 Added api/webrtcsdp.h forwarding header to work around upstream projects. by ossu · 8 years ago
  91. 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
  92. b54c63f Moving no_op_function.cc out of webrtc/build by mbonadei · 8 years ago
  93. dabbea6 Moving whitespace file up by one folder by mbonadei · 8 years ago
  94. 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
  95. e6b4723 Statically linked zxing. Without dependency on libMagick by mandermo · 8 years ago
  96. 4b7c952 Reland of "Log audio network adapter decisions in event log." by minyue · 8 years ago
  97. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
  98. 62d02c3 Unit test out of band H264 SPS,PPS within RtpStreamReceiver. by johan · 8 years ago
  99. 822d258 Move webrtc/build/android -> tools-webrtc/android by mbonadei · 8 years ago
  100. 81eab61 Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo. by brandtr · 8 years ago