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gerrit-public.fairphone.software / platform / external / webrtc / a13e7a1d4663e945195966f92298085700af46e4
  1. a13e7a1 Add factory to create scalability structures by name by Danil Chapovalov · 4 years, 3 months ago
  2. 97910da Do not use internal source in H.264 bitstream rewriting tests. by Mirta Dvornicic · 4 years, 3 months ago
  3. c6801d4 sdp: parse and serialize b=TIAS by Philipp Hancke · 4 years, 3 months ago
  4. 9ad1f6f Reland "Delete PeerConnectionInterface::BitrateParameters" by Niels Möller · 4 years, 3 months ago
  5. c888ffa Delete CompositeDataChannelTransport by Niels Möller · 4 years, 3 months ago
  6. 121f32c Roll chromium_revision 812a4946f7..a29ceb7864 (788405:788510) by chromium-webrtc-autoroll · 4 years, 3 months ago
  7. 77ca290 Roll chromium_revision 20d0aa1d03..812a4946f7 (788294:788405) by chromium-webrtc-autoroll · 4 years, 3 months ago
  8. 26d41f2 Roll chromium_revision 3e578a0ddf..20d0aa1d03 (787863:788294) by chromium-webrtc-autoroll · 4 years, 3 months ago
  9. 80d2159 Use Android Q API to test if MediaCodecInfo is HW Accelerated by Byoungchan Lee · 4 years, 3 months ago
  10. 84bb634 Delete legacy cricket::RtpHeaderExtension struct as unused by Danil Chapovalov · 4 years, 3 months ago
  11. 47f87ab Roll chromium_revision 432e660d6d..3e578a0ddf (787714:787863) by chromium-webrtc-autoroll · 4 years, 3 months ago
  12. c7c412a Check for null before accessing SctpTransport map. by Taylor Brandstetter · 4 years, 3 months ago
  13. f60d4c2 Revert "Delete PeerConnectionInterface::BitrateParameters" by Artem Titov · 4 years, 3 months ago
  14. 6e4416c Roll chromium_revision f629a87d19..432e660d6d (787611:787714) by chromium-webrtc-autoroll · 4 years, 3 months ago
  15. 4ab7dde Migrate stray leftovers from rtc_base/ and test/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  16. 4379a7d Reland "Migrate modules/desktop_capture and modules/video_capture to webrtc::Mutex." by Markus Handell · 4 years, 3 months ago
  17. 30c8eaa Add CPEPrefix. by Adrian Taylor · 4 years, 3 months ago
  18. e2dfe74 Delete PeerConnectionInterface::BitrateParameters by Niels Möller · 4 years, 3 months ago
  19. 4819098 Auto roller: send trooper notifications only when Commit-Queue+2. by Jeremy Leconte · 4 years, 3 months ago
  20. 21621e9 Delete obsolete method JsepTransport::NegotiateDatagramTransport by Niels Möller · 4 years, 3 months ago
  21. acc1123 Roll chromium_revision 598b095453..f629a87d19 (787511:787611) by chromium-webrtc-autoroll · 4 years, 3 months ago
  22. 59f3c68 Inclusive language in DEPS. by Mirko Bonadei · 4 years, 3 months ago
  23. 35926d9 Revert "Test luci.notifier." by Mirko Bonadei · 4 years, 3 months ago
  24. 662918e Roll chromium_revision f0f2118569..598b095453 (787142:787511) by chromium-webrtc-autoroll · 4 years, 3 months ago
  25. d3ec791 Reland "Test luci.notifier." by Mirko Bonadei · 4 years, 3 months ago
  26. 2e978e3 Revert "Test luci.notifier." by Mirko Bonadei · 4 years, 3 months ago
  27. bd393b2 Add target_sdk_version to rtc_test. by Mirko Bonadei · 4 years, 3 months ago
  28. 84a812e Check old_vector_size prior to copying in RTPFragmentationHeader::Resize by Dan Minor · 4 years, 3 months ago
  29. 8df59bc Test luci.notifier. by Mirko Bonadei · 4 years, 3 months ago
  30. 1added5 Revert "Migrate modules/desktop_capture and modules/video_capture to webrtc::Mutex." by Markus Handell · 4 years, 3 months ago
  31. 0800010 peerconnection: remove old helper function by Philipp Hancke · 4 years, 3 months ago
  32. edacbd5 Reland "Implement packets_(sent | received) for RTCTransportStats" by Artem Titov · 4 years, 3 months ago
  33. 76ad2e0 Add jitter buffer delay metric to pc level tests by Andrey Logvin · 4 years, 3 months ago
  34. 265931e Migrate VideoStreamDecoderImpl to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  35. bc633d7 Roll chromium_revision a69b9c614c..f0f2118569 (786202:787142) by chromium-webrtc-autoroll · 4 years, 3 months ago
  36. 4e5bc9f Reland "Complete migration from "track" to "inbound-rtp" stats" by Eldar Rello · 4 years, 3 months ago
  37. 5e98742 Add missing dependencies to java targets. by Mirko Bonadei · 4 years, 3 months ago
  38. f7303e6 Migrate leftovers in media/ and modules/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  39. 85585f4 Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  40. e6f3897 Revert "Complete migration from "track" to "inbound-rtp" stats" by Zeke Chin · 4 years, 3 months ago
  41. 3a034e1 Split DataChannel into two separate classes for RTP and SCTP. by Taylor Brandstetter · 4 years, 3 months ago
  42. 61c2d99 Add skeleton of new capturer that uses Windows.Graphics.Capture API by Austin Orion · 4 years, 3 months ago
  43. 60ed459 Migrate a leftover in test/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  44. 73ecede [Adaptation] Move AdaptationConstraints to VideoStreamAdapter by Evan Shrubsole · 4 years, 3 months ago
  45. e56976d Reland "Migrate test/time_controller to webrtc::Mutex." by Markus Handell · 4 years, 3 months ago
  46. 517f81e [Adaptation] Make ResourceAdaptationProcessorInterface resources thread-safe by Evan Shrubsole · 4 years, 3 months ago
  47. 5fbd758 Revert "Build: Disable the iLBC audio codec by default" by Olga Sharonova · 4 years, 3 months ago
  48. dc4d422 [Adaptation] Move AdaptationListeners to VideoStreamAdapter by Evan Shrubsole · 4 years, 3 months ago
  49. 94fe0d3 Complete migration from "track" to "inbound-rtp" stats by Eldar Rello · 4 years, 3 months ago
  50. 18523c3 Migrate rtc_base to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  51. 44dd3d7 Migrate modules/desktop_capture and modules/video_capture to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  52. 531bd0f Clarify logging docs. by Markus Handell · 4 years, 3 months ago
  53. 563d497 SimulatedTaskQueue: release lock before destroying tasks. by Markus Handell · 4 years, 3 months ago
  54. 4465ee1 Fix 'Perf Android32 (M Nexus5)' config in MB. by Mirko Bonadei · 4 years, 3 months ago
  55. 927ee49 Reland "Add possibility to expand strings in the autoroller." by Mirko Bonadei · 4 years, 3 months ago
  56. c1b2712 Delete RtcpDemuxer as unused by Danil Chapovalov · 4 years, 3 months ago
  57. a5a4be1 Partly migrate test/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  58. 059f4f7 Cleanup setting VideoQualityTestFixtureInterface::Param struct by Danil Chapovalov · 4 years, 3 months ago
  59. 0973a55 Revert "Add possibility to expand strings in the autoroller." by Mirko Bonadei · 4 years, 3 months ago
  60. a376518 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  61. a9e1b49 Fix support for BitBuffer::ReadNonSymmetric(&value, 1) by Danil Chapovalov · 4 years, 3 months ago
  62. e88c95e [Stats] Add more rtc::Thread::ScopedDisallowBlockingCalls to getStats(). by Henrik Boström · 4 years, 3 months ago
  63. b504780 Remove unused GuardedAsyncInvoker class by Tomas Gunnarsson · 4 years, 3 months ago
  64. 122fadc Revert "Migrate test/time_controller to webrtc::Mutex." by Markus Handell · 4 years, 3 months ago
  65. bd49463 Roll chromium_revision b5359525fc..a69b9c614c (785857:786202) by chromium-webrtc-autoroll · 4 years, 3 months ago
  66. 6fafbe3 [Stats] Optimization: Minimize number of thread-invokes in getStats(). by Henrik Boström · 4 years, 3 months ago
  67. 9b35da8 Revert "Implement packets_(sent | received) for RTCTransportStats" by Mirko Bonadei · 4 years, 3 months ago
  68. 5f61282 Migrate modules/audio_device to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  69. adbfd1d VideoAnalyzer: remove lock recursions. by Markus Handell · 4 years, 3 months ago
  70. 6cc893a Migrate test/pc/e2e to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  71. bfb6124 Add possibility to expand strings in the autoroller. by Mirko Bonadei · 4 years, 3 months ago
  72. 52fd96f Migrate test/time_controller to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  73. 9c96250 Migrate modules/congestion_controller to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  74. 656efbe Fix thread usage in PC level tests for getting to the IceConnected state by Artem Titov · 4 years, 3 months ago
  75. ce1ff6f Logging: reduce locking frequency by fixing TODO. by Markus Handell · 4 years, 3 months ago
  76. a827a30 Revert "Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex." by Markus Handell · 4 years, 3 months ago
  77. 462b5a6 Roll chromium_revision fa6a961223..b5359525fc (785707:785857) by chromium-webrtc-autoroll · 4 years, 3 months ago
  78. 6fcd0f8 Migrate pc/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  79. 0eba415 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  80. edcb907 Migrate modules/remote_bitrate_estimator to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  81. 9bbff07 Migrate video/adaptation and video/end_to_end_tests to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  82. 101750a Roll chromium_revision 5f083e73a6..fa6a961223 (785559:785707) by chromium-webrtc-autoroll · 4 years, 3 months ago
  83. 0df0fae Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  84. 1e257ca Migrate media/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  85. 06d034f Migrate common_video/ and examples/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  86. 9030994 Update default max nr of packets to 200. by Ivo Creusen · 4 years, 3 months ago
  87. e7c015e Migrate modules/rtp_rtcp to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  88. 6deec38 Migrate modules/video_coding to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  89. fb6f975 Implement packets_(sent | received) for RTCTransportStats by Artem Titov · 4 years, 3 months ago
  90. 3444a49 [Adaptation] Remove deg.prefs from ResourceAdaptationProcessor by Evan Shrubsole · 4 years, 3 months ago
  91. c212873 Relanding: Fix data channel message integrity violation by Taylor Brandstetter · 4 years, 3 months ago
  92. 48f20f8 Roll chromium_revision 31d34f548e..5f083e73a6 (785408:785559) by chromium-webrtc-autoroll · 4 years, 3 months ago
  93. 0ef4a24 Add simulated time support for PC level test. by Artem Titov · 4 years, 3 months ago
  94. a166a35 webrtc::AudioSendStream: Add lock annotation to audio_level_ by Sam Zackrisson · 4 years, 3 months ago
  95. b3a6816 Roll chromium_revision a9cc4f4732..31d34f548e (785286:785408) by chromium-webrtc-autoroll · 4 years, 3 months ago
  96. 8fe932a Migrate call/ to webrtc::Mutex. by Markus Handell · 4 years, 3 months ago
  97. 6287280 Migrate audio/ to use webrtc::Mutex by Markus Handell · 4 years, 3 months ago
  98. afd1dcb [Adaptation] Move TriggerAdaptationDueToFrameDroppedDueToSize by Evan Shrubsole · 4 years, 3 months ago
  99. c3efe1a Inclusive language in //rtc_base. by Mirko Bonadei · 4 years, 3 months ago
  100. c7f0dff Convert GN libs lists to frameworks by Sylvain Defresne · 4 years, 3 months ago