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gerrit-public.fairphone.software / platform / external / webrtc / aea1a017edcad97c63c80d21973c7bf8522f5247
  1. aea1a01 Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS by magjed · 7 years ago
  2. 3c9151b Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ ) by mbonadei · 7 years ago
  3. a62a82b Creating libwebrtc bundle jar by mbonadei · 7 years ago
  4. fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 7 years ago
  5. 2d4d653 Fix msan flake in rtcstats_integrationtest.cc. by hbos · 7 years ago
  6. c854ac3 Stop camera onStop instead of onPause. by sakal · 7 years ago
  7. 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 7 years ago
  8. 7319f26 Roll chromium_revision 780d18a4ff..7649e76842 (445004:445027) by buildbot · 7 years ago
  9. 30fe5e0 Prevent downstream linter warnings. by sakal · 7 years ago
  10. 3556406 Camera1Session: Fix camera sometimes getting stopped twice. by sakal · 7 years ago
  11. 9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 7 years ago
  12. fd6c94d Allow more config changes for CallActivity. by sakal · 7 years ago
  13. 3e92290 Load library dependencies in AppRTCMobile. by sakal · 7 years ago
  14. be850e1 Clear out cached codecs when calculating new codec lists. by noahric · 7 years ago
  15. 204030a Roll chromium_revision bdeae63b37..780d18a4ff (444971:445004) by buildbot · 7 years ago
  16. a01d2f5 Roll chromium_revision 34215edf2e..bdeae63b37 (444898:444971) by buildbot · 7 years ago
  17. 888874f Allow GCC 4.9 to compile Chromium by floppymaster · 7 years ago
  18. 8944ab3 Roll chromium_revision 1a7fcf6220..34215edf2e (444851:444898) by buildbot · 7 years ago
  19. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 7 years ago
  20. 9d643e8 Roll chromium_revision 113278e435..1a7fcf6220 (444801:444851) by buildbot · 7 years ago
  21. 537798b Roll chromium_revision d50ce8a895..113278e435 (444743:444801) by buildbot · 7 years ago
  22. 9410b51 GN: Add audio_conference_mixer_unittests to modules_unittests. by ehmaldonado · 7 years ago
  23. d748863 Fix PseudoTcp to handle incoming packets with invalid SEQ field by sergeyu · 7 years ago
  24. 3cd896c Roll chromium_revision 6d8c754784..d50ce8a895 (444712:444743) by buildbot · 7 years ago
  25. eef94d9 Video collected by VideoFileRenderer is first saved on the native heap, then saved to disk during release. by mandermo · 7 years ago
  26. 3626865 GN: Refactor modules_unittests to eliminate package boundary violations. by ehmaldonado · 7 years ago
  27. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  28. 093dac1 Roll chromium_revision 0f65d3f753..6d8c754784 (444698:444712) by buildbot · 8 years ago
  29. b935984 Revert of Move congestion controller processing to the pacer thread. (patchset #5 id:80001 of https://codereview.webrtc.org/2637783003/ ) by nisse · 8 years ago
  30. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  31. daeffb2 Roll chromium_revision be0566f991..0f65d3f753 (444668:444698) by buildbot · 8 years ago
  32. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  33. 04926b8 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) by kjellander · 8 years ago
  34. f847178 Roll chromium_revision adc103db18..be0566f991 (444630:444668) by buildbot · 8 years ago
  35. 4b96466 Roll chromium_revision e9762587b9..adc103db18 (444575:444630) by buildbot · 8 years ago
  36. f15825f Roll chromium_revision 0bc260f9e8..e9762587b9 (444497:444575) by buildbot · 8 years ago
  37. 3078b55 Reduce the log verbosity in sslstreamadapter_unittest by skvlad · 8 years ago
  38. d1c0998 Adding OrtcFactory, and changing UdpTransport to match current plan. by deadbeef · 8 years ago
  39. 27edfbc Roll chromium_revision 10fecf4ab1..0bc260f9e8 (444374:444497) by buildbot · 8 years ago
  40. a3c8c90 Roll chromium_revision d9e076c478..10fecf4ab1 (444338:444374) by buildbot · 8 years ago
  41. d99a200 Adding some features to proxy.h, and restructuring the macros. by deadbeef · 8 years ago
  42. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  43. f20dd00 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ ) by philipel · 8 years ago
  44. 6da303d Reland of Delete rtc::linked_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2579753002/ ) by nisse · 8 years ago
  45. 1b3ce86 Roll chromium_revision 5e5d50d1fe..d9e076c478 (444317:444338) by buildbot · 8 years ago
  46. fcc6006 Clear the FrameBuffer in case of a backward jump in the picture id. by philipel · 8 years ago
  47. 44303ea Revert of Add experimental simulcast screen content mode (patchset #5 id:80001 of https://codereview.webrtc.org/2636443002/ ) by sprang · 8 years ago
  48. e8abe3e Revert of New method StatsObserver::OnCompleteReports, passing ownership. (patchset #2 id:20001 of https://codereview.webrtc.org/2584553002/ ) by nisse · 8 years ago
  49. 2f67b82 Fixing peerconnection reddish video issue by mbonadei · 8 years ago
  50. 5850a94 Add failure type parameter to onFailure callback. by sakal · 8 years ago
  51. 2fcd2dd Update YuvConverter to use GlTextureFrameBuffer. by sakal · 8 years ago
  52. a77ce78 Roll chromium_revision 5d804c8487..5e5d50d1fe (444296:444317) by buildbot · 8 years ago
  53. 4a0c764 Add rtcp::TransportFeedback::GetReceivedPackets() by danilchap · 8 years ago
  54. e0e3bdf Refactor OveruseFrameDetector to use timing in us units by nisse · 8 years ago
  55. d3fabe5 Improve computational performance of BWE by switching list to deque. by terelius · 8 years ago
  56. 1d2d789 Fix race in EndToEndTest.ReceivesFlexfecAndSendsCorrespondingRtcp. by brandtr · 8 years ago
  57. a28e971 Add experimental simulcast screen content mode by sprang · 8 years ago
  58. b3dc2b7 Move congestion controller processing to the pacer thread. by nisse · 8 years ago
  59. c0370ef Roll chromium_revision 9057f45850..5d804c8487 (444268:444296) by buildbot · 8 years ago
  60. f3d5d89 Roll chromium_revision aa99b0c20f..9057f45850 (444210:444268) by buildbot · 8 years ago
  61. c7953fa Remove the IceTransportInternal2. by zhihuang · 8 years ago
  62. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  63. b308b03 Roll chromium_revision b09bc11cad..aa99b0c20f (444155:444210) by buildbot · 8 years ago
  64. 7fa4a72 Increase bitrate adjustment values for VP8 Exynos encoder by Alex Glaznev · 8 years ago
  65. 6dbbd89 Fix BitrateProber to match the requested bitrate more precisely by sergeyu · 8 years ago
  66. f7303fc Roll chromium_revision 71ee072729..b09bc11cad (444100:444155) by buildbot · 8 years ago
  67. 2e37a72 Roll chromium_revision 5e608bb301..71ee072729 (444065:444100) by buildbot · 8 years ago
  68. 9af18bf Roll chromium_revision e44e863e19..5e608bb301 (444017:444065) by buildbot · 8 years ago
  69. e5cbc20 Android: AppRTCMobile: Don't leak CallActivity. by sakal · 8 years ago
  70. 61c98e0 Remove dependency to Chromium code from WebRTC Java code. by sakal · 8 years ago
  71. e08b253 Remove unused lambda capture to unbreak downstream code. by solenberg · 8 years ago
  72. 0b2d3e2 Revert of Fix flaky WebRtcVideoChannel2BaseTest.GetStats T (patchset #1 id:1 of https://codereview.webrtc.org/2634273002/ ) by perkj · 8 years ago
  73. 2013e29 Disable automatic scaling in tests. by nisse · 8 years ago
  74. 311a64c Fix flaky WebRtcVideoChannel2BaseTest.GetStats T by perkj · 8 years ago
  75. c08c191 Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ ) by philipel · 8 years ago
  76. 6c0fd43 Roll chromium_revision c1fcfd706a..e44e863e19 (443985:444017) by buildbot · 8 years ago
  77. e8aca24 Move file capturer/renderer tests to the correct location. by sakal · 8 years ago
  78. 0f0763d Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
  79. 804ab6f Parse MedianSlopeFilter-parameters to the correct variables. by terelius · 8 years ago
  80. 7456817 Comparison of videos with reference frame not starting from zero by mandermo · 8 years ago
  81. 160e4a7 RTCMediaStreamTrackStats.kind added and collected. by hbos · 8 years ago
  82. 9b96a17 Android GlTextureFrameBuffer: Re-attach texture in setSize by magjed · 8 years ago
  83. 1fd08c1 GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. by ehmaldonado · 8 years ago
  84. ece0571 UdpTransport:IsIpAddressValid: Added extra :: check for ipv6 by ossu · 8 years ago
  85. da5e9d0 Initiate mid-call probing even if estimated bitrate is at max configured bitrate. by philipel · 8 years ago
  86. fa5a368 Let FlexfecReceiveStreamImpl send RTCP RRs. by brandtr · 8 years ago
  87. 9506e12 Reset pendingCameraSwitch as false after failed to post switchCameraOnThread to camera thread. by hankzhang8945 · 8 years ago
  88. f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
  89. 04a057b Add missing if-clause for residual_echo_likelihood_recent_max by henrik.lundin · 8 years ago
  90. d3c3a4e Roll chromium_revision 9eb76629f8..c1fcfd706a (443880:443985) by buildbot · 8 years ago
  91. c117e2e Do not classify error after stopping the camera as a startup failure. by sakal · 8 years ago
  92. 2df0734 Add WebRTC.BWE.MidCallProbing.* metrics. by philipel · 8 years ago
  93. 00c7ad1 Roll chromium_revision 07adb5c7a6..9eb76629f8 (443862:443880) by buildbot · 8 years ago
  94. 7064d59 RTCTransportStats.dtlsState replaces .activeConnection by hbos · 8 years ago
  95. 3d200bd Remove FlexfecConfig and replace with specific struct in VideoSendStream. by brandtr · 8 years ago
  96. 57c2fff Periodically update channel parameters and send TargetBitrate message. by sprang · 8 years ago
  97. 84abeb1 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. by hbos · 8 years ago
  98. 93f16d7 delete redundant members in ViEEncoder by kthelgason · 8 years ago
  99. e78d266 Make FakeEncoder and FakeH264Encoder thread safe. by brandtr · 8 years ago
  100. 037b93a Replace default locale with US locale on Android. by sakal · 8 years ago