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gerrit-public.fairphone.software / platform / external / webrtc / e372d3c51949672de3a4236f5ff3fcc011d528eb
  1. e372d3c Add event log visualization of rtp timestamps over time. by stefan · 7 years ago
  2. a55f021 Add 120ms frame ability to ANA by michaelt · 7 years ago
  3. ed01647 Remove bad DCHECK added as part of https://codereview.webrtc.org/2452163004/ by solenberg · 7 years ago
  4. b33eed2 Fix perf issue when timinig out receiver infos in RTCP. by stefan · 7 years ago
  5. cc99bc2 Change StunMessage::AddAttribute return type from bool to void. by nisse · 7 years ago
  6. f7826d6 Remove InlinedApi lint ignore. by sakal · 7 years ago
  7. a29d5ec Make 'webrtc' target a complete static library on Linux, Android and Windows by kjellander · 7 years ago
  8. 24af663 Adding Java wrapper for DtmfSender. by deadbeef · 7 years ago
  9. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 7 years ago
  10. 2e03c66 Adding build switch for Opus that supports 120ms ptime. by minyue · 7 years ago
  11. d3d3ba5 Revert of Enable audio streams to send padding. (patchset #4 id:60001 of https://codereview.webrtc.org/2652893004/ ) by deadbeef · 7 years ago
  12. 1cbf518 Roll chromium_revision 6b2002254c..496a750d38 (447561:447619) by buildbot · 7 years ago
  13. 353e7e1 Roll chromium_revision 9f2c537112..6b2002254c (447517:447561) by buildbot · 7 years ago
  14. e35f89a Enable audio streams to send padding. by stefan · 7 years ago
  15. 46fbb7d Roll chromium_revision ccc17b815a..9f2c537112 (447493:447517) by buildbot · 7 years ago
  16. b1ca073 Rename adaptation api methods, extended vie_encoder unit test. by sprang · 7 years ago
  17. d83b967 Replace consecutive-losses count by a calculation of first-order-FEC recoverability. by elad.alon · 7 years ago
  18. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 7 years ago
  19. 77f0580 Add new step graph type to event log visualization tool. Currently used for bitrate estimate and accumulated packet count, but could in general be used for any metric that is piecewise constant. by terelius · 7 years ago
  20. a565f92 Roll chromium_revision e87481817b..ccc17b815a (447482:447493) by buildbot · 7 years ago
  21. 099110c Don't send audio packets if the network is down. by stefan · 7 years ago
  22. 4637b6a Consistent 30% improvement in audio mixer running time. by aleloi · 7 years ago
  23. 35fc2aa Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ ) by minyue · 7 years ago
  24. 2ad42ca Roll chromium_revision 8346af5a71..e87481817b (447464:447482) by buildbot · 7 years ago
  25. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 7 years ago
  26. 803dc29 Enable cpplint and fix cpplint errors in webrtc/api by oprypin · 7 years ago
  27. 83399ca Drop frames until specified bitrate is achieved. by kthelgason · 7 years ago
  28. fdd9b85 Roll chromium_revision e4d460e023..8346af5a71 (447441:447464) by buildbot · 7 years ago
  29. a1cf88d Roll chromium_revision 9d90548426..e4d460e023 (447390:447441) by buildbot · 7 years ago
  30. 3f6d817 Roll chromium_revision 2ed48364ed..9d90548426 (447343:447390) by buildbot · 7 years ago
  31. dc20e26 Use correct calling convention for CreateThread callback on Windows. by deadbeef · 7 years ago
  32. 3e4ebc7 Roll chromium_revision 0851a43de7..2ed48364ed (447237:447343) by buildbot · 7 years ago
  33. ac61b74 Refactor FakeAudioDevice to have separate methods for starting recording and playout. by perkj · 7 years ago
  34. 1c05625 Fix race condition in FrameBuffer2 by philipel · 7 years ago
  35. 54340d8 Change opus min bitrate. by michaelt · 7 years ago
  36. cf34fde Roll chromium_revision 721746ebca..0851a43de7 (447221:447237) by buildbot · 7 years ago
  37. 7f08e82 Fix per regression in probing. by stefan · 7 years ago
  38. 6fb4f56 Reland of move usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2666103002/ ) by oprypin · 7 years ago
  39. d1685ab Revert of Remove usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2660823003/ ) by oprypin · 7 years ago
  40. 0fe1216 Move more calls to webrtc::field_trial::FindFullName into ctor, thereby minimizing the non-trivial cost of repeated string comparisons. by elad.alon · 7 years ago
  41. 89f281c Roll chromium_revision f74de5a3c9..721746ebca (447212:447221) by buildbot · 7 years ago
  42. b2caab7 Remove usage of deprecated g_type_init API by oprypin · 7 years ago
  43. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 7 years ago
  44. 63b14b7 Add override declarations to PeerConnectionObserver subclasses, and delete obsolete methods. by nisse · 7 years ago
  45. 1783f16 Roll chromium_revision a2c4dd1ab5..f74de5a3c9 (447201:447212) by buildbot · 7 years ago
  46. a7ee14e Suppress Memcheck:Uninitialized error when printing rtc::optional. by philipel · 7 years ago
  47. 1e4e8cb Add CreatePeerConnectionFactory overloads that take audio codec factory args by kwiberg · 7 years ago
  48. 7ce109a Replace the easy cases of VERIFY usage. by nisse · 7 years ago
  49. 96a9fa0 Removing webrtc/build folder by mbonadei · 7 years ago
  50. 9f9a1c7 Roll chromium_revision e7b7b06987..a2c4dd1ab5 (447179:447201) by buildbot · 7 years ago
  51. a4def99 Roll chromium_revision 3eee970eb6..e7b7b06987 (447079:447179) by buildbot · 7 years ago
  52. 02839ae Roll chromium_revision 5555191b32..3eee970eb6 (447020:447079) by buildbot · 7 years ago
  53. c14b7ed iSAC float decoder: Don't read past end of initialized part of buffer by kwiberg · 7 years ago
  54. ba3f411 Roll chromium_revision 716b1b3275..5555191b32 (446992:447020) by buildbot · 7 years ago
  55. a6a6d65 Instantly pass network changes to controllers in audio network adaptor. by minyue · 7 years ago
  56. 7b8cddd Roll chromium_revision d7ee7cd5fa..716b1b3275 (446974:446992) by buildbot · 7 years ago
  57. 4c9b4af Compute packet loss for event log visualization similar to how it is defined in RFC 3550. by terelius · 7 years ago
  58. aa4b077 Simplify IsFmtpParam according to RFC 4855. by ossu · 7 years ago
  59. 55d6539 Roll chromium_revision 4f0acca4ba..d7ee7cd5fa (446964:446974) by buildbot · 7 years ago
  60. a6ca518 iSAC: Untangle some cyclic dependencies by kwiberg · 7 years ago
  61. 4fb9746 Add presubmit check to prevent package boundary violations. by ehmaldonado · 7 years ago
  62. 9cbb0a1 Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) by ehmaldonado · 7 years ago
  63. 26d79ee Roll chromium_revision d2bee43df5..4f0acca4ba (446960:446964) by buildbot · 7 years ago
  64. 1c0dea8 Delete VideoFrame::set_render_time_ms. by nisse · 7 years ago
  65. a26330a Only define NO_RETURN if undefined by agouaillard · 7 years ago
  66. 2e60484 Roll chromium_revision 169ed39de4..d2bee43df5 (446956:446960) by buildbot · 7 years ago
  67. 6f873be Roll chromium_revision b84d9d8be2..169ed39de4 (446949:446956) by buildbot · 7 years ago
  68. e9fc18a Roll chromium_revision a297e6f4d1..b84d9d8be2 (446947:446949) by buildbot · 7 years ago
  69. 93f01be Android AppRTCMobile: Fix SDP video codec reordering for multiple H264 profiles by magjed · 7 years ago
  70. c7d928b Roll chromium_revision 420b8aefb8..a297e6f4d1 (446943:446947) by buildbot · 7 years ago
  71. 467e032 Roll chromium_revision 88a4e827ea..420b8aefb8 (446940:446943) by buildbot · 7 years ago
  72. 2f83d18 Roll chromium_revision 5810aac4a8..88a4e827ea (446939:446940) by buildbot · 7 years ago
  73. bd26ba7 Only update VCMTiming on every received frame instead of every received packet. by philipel · 7 years ago
  74. 0e86529 Roll chromium_revision 73f8d7ec73..5810aac4a8 (446937:446939) by buildbot · 7 years ago
  75. cb6aef2 Roll chromium_revision a0b3e8c6b2..73f8d7ec73 (446933:446937) by buildbot · 7 years ago
  76. ae23181 Roll chromium_revision bfd4f2991d..a0b3e8c6b2 (446928:446933) by buildbot · 7 years ago
  77. 2bc3c75 Roll chromium_revision 549738ba7a..bfd4f2991d (446923:446928) by buildbot · 7 years ago
  78. 68ede36 Roll chromium_revision cff6288fd9..549738ba7a (446921:446923) by buildbot · 7 years ago
  79. cdc2894 Roll chromium_revision 90cdf58449..cff6288fd9 (446920:446921) by buildbot · 7 years ago
  80. 6a31ee8 Roll chromium_revision b2f66c7a95..90cdf58449 (446919:446920) by buildbot · 7 years ago
  81. 66d46ae Roll chromium_revision e3bc84e363..b2f66c7a95 (446911:446919) by buildbot · 7 years ago
  82. b409e23 Roll chromium_revision d1351ea096..e3bc84e363 (446900:446911) by buildbot · 7 years ago
  83. 286299d Roll chromium_revision 647709aaba..d1351ea096 (446860:446900) by buildbot · 7 years ago
  84. 4460e7f Roll chromium_revision 14ab9f9226..647709aaba (446784:446860) by buildbot · 7 years ago
  85. 7d4a327 Roll chromium_revision 22ab374ddc..14ab9f9226 (446723:446784) by buildbot · 7 years ago
  86. 0c1d060 Enable Android H264 High profile decoder by glaznev · 7 years ago
  87. 62a5dd2 Roll chromium_revision 9ff019ad14..22ab374ddc (446676:446723) by buildbot · 7 years ago
  88. 869c30f Roll chromium_revision 087876b708..9ff019ad14 (446653:446676) by buildbot · 7 years ago
  89. 16b0221 Prioritize video packets when sending padding or preemptive retransmits. by stefan · 7 years ago
  90. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 7 years ago
  91. 5a2c506 Set the start bitrate to the delay-based BWE. by stefan · 7 years ago
  92. b0ae920 RTCRTPStreamStats.mediaTrackId renamed to trackId. by hbos · 7 years ago
  93. 55d1ebb Enable periodic bitrate probing when application limited for audio BWE. by stefan · 7 years ago
  94. 206b0d7 Roll chromium_revision 6f11aa45a2..087876b708 (446630:446653) by buildbot · 7 years ago
  95. b621c3f Move Android tests under sdk/android. by sakal · 7 years ago
  96. 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
  97. 986012d iOS AppRTCMobile: Enable H264 High profile by magjed · 8 years ago
  98. 89da160 Disable flaky test VideoSendStreamTest.RemoveOverheadFromBandwidth. by aleloi · 8 years ago
  99. 69221db Adding second layer of the echo canceller 3 functionality. by peah · 8 years ago
  100. 270048c Roll chromium_revision a429302c3d..6f11aa45a2 (446615:446630) by buildbot · 8 years ago