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gerrit-public.fairphone.software / platform / external / webrtc / 70870b921148fdc10d1131958a03d45f0f83f6fa
  1. 70870b9 Refactor webrtc/modules/desktop_capture for GN check by mbonadei · 8 years ago
  2. bb0d56a Roll chromium_revision bd4620dd0a..4bca52942c (440003:440045) by buildbot · 8 years ago
  3. 8885150 Roll chromium_revision 306ab8027c..bd4620dd0a (439956:440003) by buildbot · 8 years ago
  4. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  5. 66f6c72 Roll chromium_revision 438fa433d3..306ab8027c (439886:439956) by buildbot · 8 years ago
  6. 494dff4 Fix a screen capture issue on retina macOS devices. by erikchen · 8 years ago
  7. 14a588a Roll chromium_revision 8557f6fd65..438fa433d3 (439808:439886) by buildbot · 8 years ago
  8. 1b08dc3 To verify the upcoming code changes it is required by peah · 8 years ago
  9. 0838327 Add method needed to extract frame capture and arrival timestamps from rtc event logs. by stefan · 8 years ago
  10. f7969df Roll chromium_revision 374735e0c4..8557f6fd65 (439777:439808) by buildbot · 8 years ago
  11. 64427e5 Add back video_replay. Disappeared in the gn conversion. by stefan · 8 years ago
  12. 235d5cc Fixing relative paths and adding a docstring by mbonadei · 8 years ago
  13. 4f6296f Roll chromium_revision 03cbb9aff4..374735e0c4 (439759:439777) by buildbot · 8 years ago
  14. a3f2d30 Remove media/base header files from rtc_media target by magjed · 8 years ago
  15. 3061276 Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros. by nisse · 8 years ago
  16. 022b54e Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker. by philipel · 8 years ago
  17. ab2ffa3 Parse FlexFEC RTP headers in Call and add integration with BWE. by brandtr · 8 years ago
  18. b36ee8d New method StatsObserver::OnCompleteReports, passing ownership. by nisse · 8 years ago
  19. 5206667 Delete unused method PayloadRouter::MaxPayloadLength. by nisse · 8 years ago
  20. 8bab796 Style cleanup in RTCPReceiver by danilchap · 8 years ago
  21. cb29be5 Roll chromium_revision 722be6f390..03cbb9aff4 (439731:439759) by buildbot · 8 years ago
  22. 2037fa4 setup_links.py: Always link to base + cleanup. by Henrik Kjellander · 8 years ago
  23. 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
  24. 03ab4e7 Update .gitignore for changes in third_party by Henrik Kjellander · 8 years ago
  25. 0012d18 Move last of third_party dirs into DEPS entries by kjellander · 8 years ago
  26. 59dbfe6 Add a unit test for Opus complexity adaptation by henrik.lundin · 8 years ago
  27. babbc30 Roll chromium_revision fab77db556..722be6f390 (439696:439731) by buildbot · 8 years ago
  28. 5cef469 Remove tools/.gitignore by Henrik Kjellander · 8 years ago
  29. a2ccfc8 Remove last files in tools/ by Henrik Kjellander · 8 years ago
  30. c18cb31 Remove tools/gritsettings by Henrik Kjellander · 8 years ago
  31. 9547033 Move tools/{msan,ubsan} -> tools-webrtc/ by Henrik Kjellander · 8 years ago
  32. 19d0489 Roll chromium_revision c6596482a6..fab77db556 (439640:439696) by buildbot · 8 years ago
  33. a4215a3 Roll chromium_revision 234bb451a0..c6596482a6 (439579:439640) by buildbot · 8 years ago
  34. d5c2d93 Roll chromium_revision 357396ffe3..234bb451a0 (439510:439579) by buildbot · 8 years ago
  35. 31cc110 Create resources/.gitignore file. by kjellander · 8 years ago
  36. 86abd6f Add an abstract class for IceTransport by zhihuang · 8 years ago
  37. 627188d Roll chromium_revision 5b0d1b3eef..357396ffe3 (439480:439510) by buildbot · 8 years ago
  38. 8b5c345 Add GUARDED_BY's in FlexfecReceiver. by brandtr · 8 years ago
  39. bb7066f Clean up storage of FlexFEC payload type in webrtc::VideoCodecSettings. by brandtr · 8 years ago
  40. 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
  41. 7472dc3 Removed undefined method from webrtcsession.h. by hbos · 8 years ago
  42. efde908 Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. by johan · 8 years ago
  43. 1865a15 Roll chromium_revision ceea172d09..5b0d1b3eef (439469:439480) by buildbot · 8 years ago
  44. 8a0d357 Roll chromium_revision 5e0dca78b3..ceea172d09 (438769:439469) by buildbot · 8 years ago
  45. 66d4b37 Move histogram for number of pause events to per stream: by asapersson · 8 years ago
  46. 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
  47. 8c1dd8d Re-enable Opus complexity tests on Android by henrik.lundin · 8 years ago
  48. 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
  49. 5bf4e08 CodecInst operator<< by kwiberg · 8 years ago
  50. 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
  51. 696c9c6 Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest. by brandtr · 8 years ago
  52. b78306a Fix segfault when PeerConnection is destroyed during stats collection. by hbos · 8 years ago
  53. 0c8c538 Properly report number of quality downscales in stats. by kthelgason · 8 years ago
  54. df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
  55. 20d05a9 Now expect the correct number of streams in EndToEndTest.GetStats. by philipel · 8 years ago
  56. b0f04fd Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  57. b6c456b Add QP stats to the statsview in AppRTCMobile for ios. by denicija · 8 years ago
  58. c3e1cab Revert of Re-enable Opus complexity tests on Android (patchset #1 id:1 of https://codereview.webrtc.org/2589673002/ ) by henrik.lundin · 8 years ago
  59. 14479dd Re-enable Opus complexity tests on Android by henrik.lundin · 8 years ago
  60. 7250b39 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  61. 9d7ea09 Revert of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2477663002/ ) by asapersson · 8 years ago
  62. b2d5577 Move tools/mb -> tools-webrtc/mb by Henrik Kjellander · 8 years ago
  63. b4ad603 Put iOS H264 High profile under a field trial by magjed · 8 years ago
  64. afd5494 Move tools/valgrind-webrtc -> tools-webrtc/valgrind by kjellander · 8 years ago
  65. 8d8816c CQ config: remove unused hide_ref_in_committed_msg. by tandrii · 8 years ago
  66. ffb865f Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ ) by skvlad · 8 years ago
  67. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  68. fb6ad3b Add full stack tests: by asapersson · 8 years ago
  69. 5bf9bed Remove redundant local variable from OveruseDetector::Detect. by terelius · 8 years ago
  70. b3df385 Remove device HW id -> marketing name mapping table for iOS devices. by kthelgason · 8 years ago
  71. 617ca31 Delete unused code from systeminfo. by kthelgason · 8 years ago
  72. ffecbbf Fix for integer overflow in NetEq. by ivoc · 8 years ago
  73. 70f39a3 In RtpPacket do not keep pointer to RtpHeaderExtensionMap by danilchap · 8 years ago
  74. 7e70fe2 Fix wrong log message. by kthelgason · 8 years ago
  75. 51813b3 Use NtpTime in RTCPSender::RtcpContext instead of pair of uint32_t by danilchap · 8 years ago
  76. eb538fd Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
  77. beafee3 Move ios_helpers to sdk folder by kthelgason · 8 years ago
  78. 8d66245 Adding Åsa and Erik as video owners. by mflodman · 8 years ago
  79. 528ec3d Don't report packets with id -1 to the transport feedback adapter as they provide no value. by stefan · 8 years ago
  80. 05d6e26 Initialize packetization mode in VideoToolbox by kthelgason · 8 years ago
  81. 9a27205 whitespace by manual git cl land. by Andrii Shyshkalov · 8 years ago
  82. 2d897bd CQ whitespace. by tandrii · 8 years ago
  83. 4cd6221 iOS: Add trendline filter to field trials. by tkchin · 8 years ago
  84. 5bc3945 Fix integer overflow in ProbeController. by sergeyu · 8 years ago
  85. b3564ad Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
  86. a97c5d2 Add ossu@ to OWNERS of audio/ and modules/audio_coding/ by ossu · 8 years ago
  87. d79f97b Fixing loopback video test by reconfiguring the encoder to correct size. by mflodman · 8 years ago
  88. 721d402 Create VideoReceiver with external VCMTiming object. by philipel · 8 years ago
  89. ac8d516 Improves release of allocated audio resources on Android. by henrika · 8 years ago
  90. 43c3821 Revert of Avoid precision loss in TrendlineEstimator from int64_t -> double conversion (patchset #7 id:120001 of https://codereview.webrtc.org/2577463002/ ) by terelius · 8 years ago
  91. 0bac07a Revert of Avoid precision loss in MedianSlopeEstimator from int64_t -> double conversion (patchset #3 id:40001 of https://codereview.webrtc.org/2578543002/ ) by terelius · 8 years ago
  92. ebcbcc3 Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss. by terelius · 8 years ago
  93. df2ceb8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ ) by nisse · 8 years ago
  94. bf65be5 Wire-up audio BWE with overhead. by michaelt · 8 years ago
  95. c12cbaf Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double. by terelius · 8 years ago
  96. 3168c7a Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. by hbos · 8 years ago
  97. 6a58f33 Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ ) by nisse · 8 years ago
  98. 7bf5369 RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs. by hbos · 8 years ago
  99. 06035cf Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ ) by nisse · 8 years ago
  100. 0b571a1 MB: Enable memcheck for the linux_memcheck trybot. by Henrik Kjellander · 8 years ago