Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 9aa3f0a2006dfc7278e0c93cd073d4bbb190b796
  1. 9aa3f0a Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 7 years ago
  2. b54c63f Moving no_op_function.cc out of webrtc/build by mbonadei · 7 years ago
  3. dabbea6 Moving whitespace file up by one folder by mbonadei · 7 years ago
  4. 69dc7db Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 7 years ago
  5. e6b4723 Statically linked zxing. Without dependency on libMagick by mandermo · 7 years ago
  6. 4b7c952 Reland of "Log audio network adapter decisions in event log." by minyue · 7 years ago
  7. 35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 7 years ago
  8. 62d02c3 Unit test out of band H264 SPS,PPS within RtpStreamReceiver. by johan · 7 years ago
  9. 822d258 Move webrtc/build/android -> tools-webrtc/android by mbonadei · 7 years ago
  10. 81eab61 Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo. by brandtr · 7 years ago
  11. 0608ffd Roll chromium_revision 59592eaa98..319b885718 (445345:445689) by buildbot · 7 years ago
  12. 365aebd Make CongestionController::remote_bitrate_estimator_ a non-pointer. by nisse · 7 years ago
  13. d2b092f Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. by johan · 7 years ago
  14. 15389c0 Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. by nisse · 7 years ago
  15. 568c9e7 New simulators to test BWE at low bitrates (15-50kbps range). by terelius · 7 years ago
  16. a4a7538 Android: Script for building libwebrtc.aar. by sakal · 7 years ago
  17. e04064d Revert of Delete unused class/template ScopedMessageData. (patchset #1 id:1 of https://codereview.webrtc.org/2652663002/ ) by aleloi · 7 years ago
  18. dc2b3f3 Delete unused class CompositeMediaEngineWithFakeVoiceEngine. by nisse · 7 years ago
  19. d83fb92 Delete unused class/template ScopedMessageData. by nisse · 7 years ago
  20. c23b0b2 Delete unused classes DesktopId and ScreencastEventCatcher. by nisse · 7 years ago
  21. ad45228 Moving get_landmines.py (build/ -> tools-webrtc/) by mbonadei · 7 years ago
  22. 2b75526 Add linux_memcheck as default trybot. by Henrik Kjellander · 7 years ago
  23. 914d49d Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ ) by kjellander · 7 years ago
  24. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 7 years ago
  25. 4c78702 iOS: Add MedianSlopeFilter field trial. by tkchin · 7 years ago
  26. 5c4f24a Move implmentation specific constants out of rtp_header_extension.h by danilchap · 7 years ago
  27. f53d737 H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. by johan · 7 years ago
  28. e1405ad Removed double-special-casing of ISAC in libjingle and WebRtcVoE. by ossu · 7 years ago
  29. cb893ee Removing unused code from webrtc/build by mbonadei · 7 years ago
  30. 1bed2e4 video_loopback: fall back to fake capturer if we can't open camera. by sprang · 7 years ago
  31. 435ddf9 Add TransportFeedbackPacketLossTracker. by minyue · 7 years ago
  32. ed582f7 Script to start stubbed loopback video test with Espresso by mandermo · 7 years ago
  33. 0ebdf27 Delete or update left-over ASSERT use and comments. by nisse · 7 years ago
  34. da25006 Fixed public_deps for libjingle_peerconnection{,_api} by ossu · 7 years ago
  35. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 7 years ago
  36. 9c3d4c4 Stop leaking FlexfecReceiveStream objects after call shutdown. by brandtr · 7 years ago
  37. a067013 Minor style change suggested by internal static analysis tool. by aleloi · 7 years ago
  38. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 7 years ago
  39. f49ff26 GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false by ehmaldonado · 7 years ago
  40. fd870db Add metric for decode time and max decode time in video quality tests. by philipel · 7 years ago
  41. 0112403 Minor style change suggested by internal static analysis tool. by aleloi · 7 years ago
  42. a31cdbc Roll chromium_revision dcc5978539..59592eaa98 (445328:445345) by buildbot · 7 years ago
  43. 0b56279 Catch failure to load native dependencies. by sakal · 7 years ago
  44. de8ca92 New script to count usage of C++ classes. by nisse · 7 years ago
  45. b55bd97 Reland of Creating libwebrtc bundle jar (patchset #1 id:1 of https://codereview.webrtc.org/2640023010/ ) by mbonadei · 7 years ago
  46. 5d0f2e8 Roll chromium_revision 269b6bc66e..dcc5978539 (445317:445328) by buildbot · 7 years ago
  47. c152434 Roll chromium_revision 7649e76842..269b6bc66e (445027:445317) by buildbot · 7 years ago
  48. 3e4faae Fixing memory leak in FakeTransportController. by deadbeef · 7 years ago
  49. 8662f94 Only set certificate on DTLS transport if fingerprint is found in SDP. by deadbeef · 7 years ago
  50. 2197e91 Remove dead code for GtkVideoRenderer. by pbos · 7 years ago
  51. f33491e Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ ) by deadbeef · 7 years ago
  52. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 7 years ago
  53. cd3180c PATENTS: fix reference by philipp.hancke · 7 years ago
  54. 7bcdb69 Ignore ufrag/password in "a=candidate" lines in SDP. by deadbeef · 7 years ago
  55. 0fc04b7 Finalize the support for building WebRTC library for iOS with bitcode by VladimirTechMan · 7 years ago
  56. f64941f RTCMediaStreamTrackStats.framesDecoded collected. by hbos · 7 years ago
  57. aea1a01 Move webrtc/sdk/DEPS to webrtc/sdk/objc/DEPS by magjed · 7 years ago
  58. 3c9151b Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ ) by mbonadei · 7 years ago
  59. a62a82b Creating libwebrtc bundle jar by mbonadei · 7 years ago
  60. fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 7 years ago
  61. 2d4d653 Fix msan flake in rtcstats_integrationtest.cc. by hbos · 7 years ago
  62. c854ac3 Stop camera onStop instead of onPause. by sakal · 7 years ago
  63. 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 7 years ago
  64. 7319f26 Roll chromium_revision 780d18a4ff..7649e76842 (445004:445027) by buildbot · 7 years ago
  65. 30fe5e0 Prevent downstream linter warnings. by sakal · 7 years ago
  66. 3556406 Camera1Session: Fix camera sometimes getting stopped twice. by sakal · 7 years ago
  67. 9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 7 years ago
  68. fd6c94d Allow more config changes for CallActivity. by sakal · 7 years ago
  69. 3e92290 Load library dependencies in AppRTCMobile. by sakal · 7 years ago
  70. be850e1 Clear out cached codecs when calculating new codec lists. by noahric · 7 years ago
  71. 204030a Roll chromium_revision bdeae63b37..780d18a4ff (444971:445004) by buildbot · 7 years ago
  72. a01d2f5 Roll chromium_revision 34215edf2e..bdeae63b37 (444898:444971) by buildbot · 7 years ago
  73. 888874f Allow GCC 4.9 to compile Chromium by floppymaster · 7 years ago
  74. 8944ab3 Roll chromium_revision 1a7fcf6220..34215edf2e (444851:444898) by buildbot · 7 years ago
  75. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 7 years ago
  76. 9d643e8 Roll chromium_revision 113278e435..1a7fcf6220 (444801:444851) by buildbot · 7 years ago
  77. 537798b Roll chromium_revision d50ce8a895..113278e435 (444743:444801) by buildbot · 7 years ago
  78. 9410b51 GN: Add audio_conference_mixer_unittests to modules_unittests. by ehmaldonado · 7 years ago
  79. d748863 Fix PseudoTcp to handle incoming packets with invalid SEQ field by sergeyu · 7 years ago
  80. 3cd896c Roll chromium_revision 6d8c754784..d50ce8a895 (444712:444743) by buildbot · 7 years ago
  81. eef94d9 Video collected by VideoFileRenderer is first saved on the native heap, then saved to disk during release. by mandermo · 7 years ago
  82. 3626865 GN: Refactor modules_unittests to eliminate package boundary violations. by ehmaldonado · 7 years ago
  83. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 7 years ago
  84. 093dac1 Roll chromium_revision 0f65d3f753..6d8c754784 (444698:444712) by buildbot · 7 years ago
  85. b935984 Revert of Move congestion controller processing to the pacer thread. (patchset #5 id:80001 of https://codereview.webrtc.org/2637783003/ ) by nisse · 7 years ago
  86. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 7 years ago
  87. daeffb2 Roll chromium_revision be0566f991..0f65d3f753 (444668:444698) by buildbot · 7 years ago
  88. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 7 years ago
  89. 04926b8 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) by kjellander · 7 years ago
  90. f847178 Roll chromium_revision adc103db18..be0566f991 (444630:444668) by buildbot · 7 years ago
  91. 4b96466 Roll chromium_revision e9762587b9..adc103db18 (444575:444630) by buildbot · 7 years ago
  92. f15825f Roll chromium_revision 0bc260f9e8..e9762587b9 (444497:444575) by buildbot · 7 years ago
  93. 3078b55 Reduce the log verbosity in sslstreamadapter_unittest by skvlad · 7 years ago
  94. d1c0998 Adding OrtcFactory, and changing UdpTransport to match current plan. by deadbeef · 7 years ago
  95. 27edfbc Roll chromium_revision 10fecf4ab1..0bc260f9e8 (444374:444497) by buildbot · 7 years ago
  96. a3c8c90 Roll chromium_revision d9e076c478..10fecf4ab1 (444338:444374) by buildbot · 7 years ago
  97. d99a200 Adding some features to proxy.h, and restructuring the macros. by deadbeef · 7 years ago
  98. c8ee882 Replace use of ASSERT in test code. by nisse · 7 years ago
  99. f20dd00 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ ) by philipel · 7 years ago
  100. 6da303d Reland of Delete rtc::linked_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2579753002/ ) by nisse · 7 years ago