Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / cf29dd2b30b874c47d257373e19f01f5eb3b9caa
  1. cf29dd2 Roll chromium_revision 7486e37f46..2eff7cfefd (476883:476894) by buildbot · 8 years ago
  2. b2c630b Roll chromium_revision 0b479f2338..7486e37f46 (476845:476883) by buildbot · 8 years ago
  3. 6fac429 Use correct bitrate adjustment for Exynos VP9 HW encoder. by glaznev · 8 years ago
  4. ab10a25 Roll chromium_revision 5867068aa6..0b479f2338 (476811:476845) by buildbot · 8 years ago
  5. 84ee5c6 Force keyframe for Qualcomm HW VP8 Encoder on Android L as well, reduce forced keyframe interval on Android M. by alexlau · 8 years ago
  6. e907741 Roll chromium_revision ba0a42d0d3..5867068aa6 (476741:476811) by buildbot · 8 years ago
  7. 4b97980 Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  8. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  9. 49d2f30 Roll chromium_revision 756477827b..ba0a42d0d3 (476675:476741) by buildbot · 8 years ago
  10. e5dce2b Replacing unnecessary conditional with DCHECK in OpenSSLAdapter by deadbeef · 8 years ago
  11. 9641c13 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  12. ed3b986 Fixing SSL error that occurs when underlying socket is blocked. by deadbeef · 8 years ago
  13. 25fd62b Roll chromium_revision e83e1b29bc..756477827b (476623:476675) by buildbot · 8 years ago
  14. ed9b9ff Revert of Protect new header extension by field trial experiment to allow hardcoding it in SDP (patchset #3 id:40001 of https://codereview.webrtc.org/2922683002/ ) by ilnik · 8 years ago
  15. dc2018b Disable PeerConnectionTest.testTrackRemovalAndAddition due to flakiness by sprang · 8 years ago
  16. 0d4d57f Add RTCFileVideoCapturer class. by denicija · 8 years ago
  17. f79ade1 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 8 years ago
  18. 2047db5 Roll chromium_revision 017ba8c602..e83e1b29bc (476612:476623) by buildbot · 8 years ago
  19. cafa1d6 Protect new header extension by field trial experiment to allow hardcoding it in SDP by ilnik · 8 years ago
  20. 58c742c Call VideoCapturer.initialize directly from Java. by Sami Kalliomäki · 8 years ago
  21. dbb497a SafeMin/SafeMax: Fix wrong return type when given two enum arguments by kwiberg · 8 years ago
  22. 2ff5b83 Roll chromium_revision 0538140988..017ba8c602 (476593:476612) by buildbot · 8 years ago
  23. 4a3c9f6 Prevent memory corruption by StreamId::Set by eladalon · 8 years ago
  24. 5522021 Android: Add VideoFrame class by magjed · 8 years ago
  25. 9b93203 Change all numerical string inputs to int and remove unused stderr by jansson · 8 years ago
  26. 07633bd Rename rtp_header_extension.h to rtp_header_extension_map.h by Danil Chapovalov · 8 years ago
  27. 963e332 Roll chromium_revision 54d7eb0b3a..0538140988 (476554:476593) by buildbot · 8 years ago
  28. 2c8e8a3 Overlay REMB in total bitrate graphs in visualization tool. by terelius · 8 years ago
  29. 1476a9d Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. by brandtr · 8 years ago
  30. cbd3ee3 Remove additional_target_cpus = [ "arm" ] from iOS device build configs by kjellander · 8 years ago
  31. 62cbb23 iOS: Update device testing configuration by kjellander · 8 years ago
  32. daf9a71 Roll chromium_revision 10c0414243..54d7eb0b3a (476503:476554) by buildbot · 8 years ago
  33. 8306f50 Roll chromium_revision a58bf8c922..10c0414243 (476450:476503) by buildbot · 8 years ago
  34. 754b008 Roll chromium_revision 1cd2c85e63..a58bf8c922 (476371:476450) by buildbot · 8 years ago
  35. 0d1e27f desktopCapture: scale the cursor image according to screen scale factor on OSX by braveyao · 8 years ago
  36. 8e857d1 Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device. by Tarun Chawla · 8 years ago
  37. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  38. 7d9a55b enabling `gn check` on the whole WebRTC repo by mbonadei · 8 years ago
  39. 3d4b83d Roll chromium_revision 907652b824..1cd2c85e63 (476301:476371) by buildbot · 8 years ago
  40. 3f07549 Update I420Buffer to new VideoFrameBuffer interface by magjed · 8 years ago
  41. c757293 Roll chromium_revision 3543d97ca7..907652b824 (476263:476301) by buildbot · 8 years ago
  42. d72098a Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 8 years ago
  43. 7a2862a Fix a bug in RtcEventLogSource by henrik.lundin · 8 years ago
  44. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  45. 26f833f Roll chromium_revision 92db6b2c14..3543d97ca7 (476239:476263) by buildbot · 8 years ago
  46. 3afb899 Remove passing Android context to NetworkMonitor. by Sami Kalliomäki · 8 years ago
  47. bc9ffad Adds support for dynamic buffer size handling on recording side for iOS. by henrika · 8 years ago
  48. 7926c12 Delete unneeded includes of system_wrappers/include/sleep.h by nisse · 8 years ago
  49. 5b54213 Print configured header extensions and codecs in rtc_event_log2text. by terelius · 8 years ago
  50. 2a8856c Switch from ScheduledExecutorService to ExecutorService. by Sami Kalliomäki · 8 years ago
  51. ab84272 Remove deprecation warning from JVM::Initialize with the context parameter. by Sami Kalliomäki · 8 years ago
  52. 3edccb9 Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) by mbonadei · 8 years ago
  53. 1a6f143 Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) by mbonadei · 8 years ago
  54. 11c7b70 Remove native method VideoTrack.free which doesn't exist. by Sami Kalliomäki · 8 years ago
  55. 0687530 Re-enable EndToEndTest.PictureIdStateRetainedAfterReinitingVp8 on tsan. by brandtr · 8 years ago
  56. db5bb40 Enabling `gn check` on webrtc/test by mbonadei · 8 years ago
  57. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
  58. 405f90c Fix indexing error in event log analyzer. by terelius · 8 years ago
  59. 00b16f4 In order to land https://codereview.webrtc.org/2790933002/ and due to the by alessiob · 8 years ago
  60. 57fa768 Roll chromium_revision 8c2a1a922d..92db6b2c14 (476213:476239) by buildbot · 8 years ago
  61. 91914e2 Use IsResolution(/Framerate)ScalingEnabled methods in more places. by asapersson · 8 years ago
  62. 7fcdb6d Delete class NullRtpData and function NullObjectRtpData. by nisse · 8 years ago
  63. a0bf559 Roll chromium_revision 1053faaf3d..8c2a1a922d (476178:476213) by buildbot · 8 years ago
  64. fa7afe6 Roll chromium_revision 5f9161e0e5..1053faaf3d (476117:476178) by buildbot · 8 years ago
  65. bf5a801 Definitions of video-codec name constants don't match their declarations by VladimirTechMan · 8 years ago
  66. 7b3eca9 Add media constraint-key constants for generating offers and answers by VladimirTechMan · 8 years ago
  67. dacac34 Roll chromium_revision fb9bbbcec6..5f9161e0e5 (476046:476117) by buildbot · 8 years ago
  68. ec8f9ab Roll chromium_revision f9718e6fb6..fb9bbbcec6 (475932:476046) by buildbot · 8 years ago
  69. de4db11 Support epoll in PhysicalSocketServer. by jbauch · 8 years ago
  70. 0cdb0ff Roll chromium_revision cb625841aa..f9718e6fb6 (475892:475932) by buildbot · 8 years ago
  71. 760a076 Create unit tests for RtpDemuxer by eladalon · 8 years ago
  72. 6d3e866 Fix spelling mistake in field trial key name. by Kári Tristan Helgason · 8 years ago
  73. eb80679 Disable flaky test EndToEndTest.TestFlexfecRtpStatePreservation on linux for now. by brandtr · 8 years ago
  74. be3e539 Small cleanup of rtp_rtcp testAPI tests. by nisse · 8 years ago
  75. 0a0a2f1 Add back key for AGC field trial. by Kári Tristan Helgason · 8 years ago
  76. 029ef77 Roll chromium_revision 39b0c40647..cb625841aa (475869:475892) by buildbot · 8 years ago
  77. 01d5a0b Propagate input capture time in ViEEncoder::OnFrame. by nisse · 8 years ago
  78. 8b2b9ef Use oneof in event log proto to reduce memory consumption. by terelius · 8 years ago
  79. 2c8ac1b Avoid to build most of webrtc/sdk in chromium by mbonadei · 8 years ago
  80. 2b356ae Roll chromium_revision 4927f54d0e..39b0c40647 (475836:475869) by buildbot · 8 years ago
  81. 108904c Use overloaded UpdateNetworkMetrics() by eladalon · 8 years ago
  82. 8fdf957 Allow FPS reduction to be applied to frame listeners. by sakal · 8 years ago
  83. 76e62b0 Address some violations of chromium-style. by nisse · 8 years ago
  84. 2108326 Delete Filesystem::IterateDirectory and Filesystem::OpenFile. by nisse · 8 years ago
  85. 8fbc765 Ensure the RtcEventLog parser is backwards compatible by terelius · 8 years ago
  86. f93f5a4 Re-add removed sdk headers to the framework target. by Kári Tristan Helgason · 8 years ago
  87. 10cf684 Roll chromium_revision 7ef7f57bb6..4927f54d0e (475783:475836) by buildbot · 8 years ago
  88. 6412e4c Drop the rtp_rtcp module's dependency on call. by nisse · 8 years ago
  89. 83cd7f3 Roll chromium_revision a3671ded59..7ef7f57bb6 (475709:475783) by buildbot · 8 years ago
  90. f56923b Roll chromium_revision 9a6a2ee988..a3671ded59 (475612:475709) by buildbot · 8 years ago
  91. cff9dfd FecControler disables FEC when *below* threshold by eladalon · 8 years ago
  92. 7ec573a Roll chromium_revision aa76e566ca..9a6a2ee988 (475541:475612) by buildbot · 8 years ago
  93. c41d0c4 Have audio_device only depend on X11 if use_x11 is set. by kjellander · 8 years ago
  94. b5f5bdb Add unit tests for qp parser. by jianj · 8 years ago
  95. 6c4bbfa Update max TL0 frame interval for screensharing. by sprang · 8 years ago
  96. 9aa5c2c Roll chromium_revision 42f1569588..aa76e566ca (475502:475541) by buildbot · 8 years ago
  97. 12de077 Avoiding check on reordering configs in ANA if there is no controller has scoring point. by minyue · 8 years ago
  98. d4d2ece Change from stderr to output, add new line and replace return with break. by jansson · 8 years ago
  99. 678d2ee Cosmetic change in ANA: renaming characteristic points for consistency. by minyue · 8 years ago
  100. 07502d4 Fixing absolute path that was breaking chromium by mbonadei · 8 years ago