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gerrit-public.fairphone.software / platform / external / webrtc / f3115a043e40dfc5ccf599321de329d7d8ac1c33
  1. f3115a0 Update Valgrind location after we stopped checking out Chromium. by Henrik Kjellander · 8 years ago
  2. 1bd7ac5 Roll chromium_revision 7ebd295d2c..2fc22f5447 (440631:440646) by buildbot · 8 years ago
  3. 23b9075 Roll chromium_revision 83e9a66e23..7ebd295d2c (440620:440631) by buildbot · 8 years ago
  4. d22a35b Roll chromium_revision c51fda8eb0..83e9a66e23 (440612:440620) by buildbot · 8 years ago
  5. acce2f3 Roll chromium_revision dea296de1b..c51fda8eb0 (440586:440612) by buildbot · 8 years ago
  6. d01ed1f Fix an error in Audio Network Adaptor: time constant passed wrong. by minyue · 8 years ago
  7. e97389c If network enumeration fails, try binding to the "ANY" address. by deadbeef · 8 years ago
  8. 5222963 Roll chromium_revision 4d12661897..dea296de1b (440512:440586) by buildbot · 8 years ago
  9. 335659a Roll chromium_revision e6b29f7805..4d12661897 (440349:440512) by buildbot · 8 years ago
  10. 40610e2 Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. by deadbeef · 8 years ago
  11. fe50b4d Make class of static functions in rtp_to_ntp.h: - UpdateRtcpList - RtpToNtp by asapersson · 8 years ago
  12. bf5f529 Disable flaky VideoSendStreamTest.RemoveOverheadeFromBandwidth by danilchap · 8 years ago
  13. ebafdc8 Refactor webrtc/modules/rtp_rtcp for GN check by mbonadei · 8 years ago
  14. 000d163 Refactor webrtc/modules/audio_conference_mixer for GN check by mbonadei · 8 years ago
  15. fb78c3e Convert CRLF to unix newlines in resources/audio_coding/READ.ME by Henrik Kjellander · 8 years ago
  16. 0de11aa Landmine to clobber failing Android x86/x64 builds by Henrik Kjellander · 8 years ago
  17. 0779e8f Add copy of src/ios/build/bots/scripts to unbreak iOS Simulator bots. by Henrik Kjellander · 8 years ago
  18. 49d2f26 Update Valgrind location after we stopped checking out Chromium. by Henrik Kjellander · 8 years ago
  19. 177567c DEPS: Sync Git subtree mirrors instead of symlinking into chromium/src by kjellander@webrtc.org · 8 years ago
  20. 5262487 Disables AudioDeviceTest.StartStopPlayout on iOS by henrika · 8 years ago
  21. 8d56088 Do not call OnDecoderTiming before timing values are set. by asapersson · 8 years ago
  22. c37ad49 Revert of Make P2PTransportChannel inherit from IceTransportInternal. (patchset #3 id:80001 of https://codereview.webrtc.org/2590063002/ ) by kjellander · 8 years ago
  23. d943c48 Revert of Refactor webrtc/modules/desktop_capture for GN check (patchset #1 id:1 of https://codereview.webrtc.org/2593713002/ ) by kjellander · 8 years ago
  24. 444f170 Roll chromium_revision 47684d7e23..e6b29f7805 (440330:440349) by buildbot · 8 years ago
  25. f7881b4 Roll chromium_revision 49c85e01a1..47684d7e23 (440284:440330) by buildbot · 8 years ago
  26. d1e60d3 Roll chromium_revision 058da9559d..49c85e01a1 (440228:440284) by buildbot · 8 years ago
  27. 53c9b2f Roll chromium_revision bdbfd895a9..058da9559d (440166:440228) by buildbot · 8 years ago
  28. dcccda7 Created a java wrapper for the callback OnAddTrack to PeerConnection.Observer by zhihuang · 8 years ago
  29. b820f9c Roll chromium_revision c26bcdb5fe..bdbfd895a9 (440109:440166) by buildbot · 8 years ago
  30. 12749d8 Make P2PTransportChannel inherit from IceTransportInternal. by zhihuang · 8 years ago
  31. eec9909 Roll chromium_revision 645df3089e..c26bcdb5fe (440075:440109) by buildbot · 8 years ago
  32. b29e652 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )" by brandtr · 8 years ago
  33. 9ba94ba Roll chromium_revision 40ccb25811..645df3089e (440063:440075) by buildbot · 8 years ago
  34. 8ed8aca Readd third_party/llvm-build to .gitignore by Henrik Kjellander · 8 years ago
  35. 23368e1 RTCStatsCollectorTest: ExpectReportContainsCertificateInfo /w EXPECT_EQ by hbos · 8 years ago
  36. c42ba32 RTCStatsCollectorTest: Remove ExpectReportContainsCandidate. by hbos · 8 years ago
  37. e55b16c Drop unneeded include of media_file.h. by nisse · 8 years ago
  38. 504b95e Avoid creating receiver_time outliers in the VideoAnalyzer. by brandtr · 8 years ago
  39. f9891d5 Roll chromium_revision 4bca52942c..40ccb25811 (440045:440063) by buildbot · 8 years ago
  40. d39e16a Revert of Refactor webrtc/modules/video_processing for GN check (patchset #3 id:40001 of https://codereview.webrtc.org/2595543002/ ) by mbonadei · 8 years ago
  41. 00a810b Refactor webrtc/modules/video_processing for GN check by mbonadei · 8 years ago
  42. dbb64d8 RTCStatsCollectorTest: Remove ExpectReportContainsDataChannel. by hbos · 8 years ago
  43. 02d2a92 RTCStatsReport::AddStats DCHECKs that the ID is unique. by hbos · 8 years ago
  44. 2a495ca Refactor webrtc/modules/pacing for GN check by mbonadei · 8 years ago
  45. 01c7150 Move nat-related code to target rtc_base_tests_utils. by nisse · 8 years ago
  46. 70e4053 Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ ) by brandtr · 8 years ago
  47. ba96730 Refactor webrtc/modules/media_file for GN check by mbonadei · 8 years ago
  48. 70870b9 Refactor webrtc/modules/desktop_capture for GN check by mbonadei · 8 years ago
  49. bb0d56a Roll chromium_revision bd4620dd0a..4bca52942c (440003:440045) by buildbot · 8 years ago
  50. 8885150 Roll chromium_revision 306ab8027c..bd4620dd0a (439956:440003) by buildbot · 8 years ago
  51. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  52. 66f6c72 Roll chromium_revision 438fa433d3..306ab8027c (439886:439956) by buildbot · 8 years ago
  53. 494dff4 Fix a screen capture issue on retina macOS devices. by erikchen · 8 years ago
  54. 14a588a Roll chromium_revision 8557f6fd65..438fa433d3 (439808:439886) by buildbot · 8 years ago
  55. 1b08dc3 To verify the upcoming code changes it is required by peah · 8 years ago
  56. 0838327 Add method needed to extract frame capture and arrival timestamps from rtc event logs. by stefan · 8 years ago
  57. f7969df Roll chromium_revision 374735e0c4..8557f6fd65 (439777:439808) by buildbot · 8 years ago
  58. 64427e5 Add back video_replay. Disappeared in the gn conversion. by stefan · 8 years ago
  59. 235d5cc Fixing relative paths and adding a docstring by mbonadei · 8 years ago
  60. 4f6296f Roll chromium_revision 03cbb9aff4..374735e0c4 (439759:439777) by buildbot · 8 years ago
  61. a3f2d30 Remove media/base header files from rtc_media target by magjed · 8 years ago
  62. 3061276 Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros. by nisse · 8 years ago
  63. 022b54e Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker. by philipel · 8 years ago
  64. ab2ffa3 Parse FlexFEC RTP headers in Call and add integration with BWE. by brandtr · 8 years ago
  65. b36ee8d New method StatsObserver::OnCompleteReports, passing ownership. by nisse · 8 years ago
  66. 5206667 Delete unused method PayloadRouter::MaxPayloadLength. by nisse · 8 years ago
  67. 8bab796 Style cleanup in RTCPReceiver by danilchap · 8 years ago
  68. cb29be5 Roll chromium_revision 722be6f390..03cbb9aff4 (439731:439759) by buildbot · 8 years ago
  69. 2037fa4 setup_links.py: Always link to base + cleanup. by Henrik Kjellander · 8 years ago
  70. d5236e2 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ ) by magjed · 8 years ago
  71. 03ab4e7 Update .gitignore for changes in third_party by Henrik Kjellander · 8 years ago
  72. 0012d18 Move last of third_party dirs into DEPS entries by kjellander · 8 years ago
  73. 59dbfe6 Add a unit test for Opus complexity adaptation by henrik.lundin · 8 years ago
  74. babbc30 Roll chromium_revision fab77db556..722be6f390 (439696:439731) by buildbot · 8 years ago
  75. 5cef469 Remove tools/.gitignore by Henrik Kjellander · 8 years ago
  76. a2ccfc8 Remove last files in tools/ by Henrik Kjellander · 8 years ago
  77. c18cb31 Remove tools/gritsettings by Henrik Kjellander · 8 years ago
  78. 9547033 Move tools/{msan,ubsan} -> tools-webrtc/ by Henrik Kjellander · 8 years ago
  79. 19d0489 Roll chromium_revision c6596482a6..fab77db556 (439640:439696) by buildbot · 8 years ago
  80. a4215a3 Roll chromium_revision 234bb451a0..c6596482a6 (439579:439640) by buildbot · 8 years ago
  81. d5c2d93 Roll chromium_revision 357396ffe3..234bb451a0 (439510:439579) by buildbot · 8 years ago
  82. 31cc110 Create resources/.gitignore file. by kjellander · 8 years ago
  83. 86abd6f Add an abstract class for IceTransport by zhihuang · 8 years ago
  84. 627188d Roll chromium_revision 5b0d1b3eef..357396ffe3 (439480:439510) by buildbot · 8 years ago
  85. 8b5c345 Add GUARDED_BY's in FlexfecReceiver. by brandtr · 8 years ago
  86. bb7066f Clean up storage of FlexFEC payload type in webrtc::VideoCodecSettings. by brandtr · 8 years ago
  87. 0ad2111 Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ ) by danilchap · 8 years ago
  88. 7472dc3 Removed undefined method from webrtcsession.h. by hbos · 8 years ago
  89. efde908 Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. by johan · 8 years ago
  90. 1865a15 Roll chromium_revision ceea172d09..5b0d1b3eef (439469:439480) by buildbot · 8 years ago
  91. 8a0d357 Roll chromium_revision 5e0dca78b3..ceea172d09 (438769:439469) by buildbot · 8 years ago
  92. 66d4b37 Move histogram for number of pause events to per stream: by asapersson · 8 years ago
  93. 0cd27ba Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ ) by kthelgason · 8 years ago
  94. 8c1dd8d Re-enable Opus complexity tests on Android by henrik.lundin · 8 years ago
  95. fe04bd4 Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ ) by kthelgason · 8 years ago
  96. 5bf4e08 CodecInst operator<< by kwiberg · 8 years ago
  97. 35b41a2 Reland of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2585183002/ ) by asapersson · 8 years ago
  98. 696c9c6 Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest. by brandtr · 8 years ago
  99. b78306a Fix segfault when PeerConnection is destroyed during stats collection. by hbos · 8 years ago
  100. 0c8c538 Properly report number of quality downscales in stats. by kthelgason · 8 years ago