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gerrit-public.fairphone.software / platform / external / webrtc / 501bfba0cb5f9649956451dd0d14f56b62af9af4
  1. 501bfba Split rtp_receiver for readability. by Ruslan Burakov · 6 years ago
  2. b66003c Delete video source proxying in WebRtcVideoSendStream by Niels Möller · 6 years ago
  3. 6df89cc Revert "Partial frame capture API part 2" by Ilya Nikolaevskiy · 6 years ago
  4. b00eb19 Removes Start/Stop on network emulation manager. by Sebastian Jansson · 6 years ago
  5. eb7589e Revert "Partial frame capture API part 3" by Ilya Nikolaevskiy · 6 years ago
  6. fd5d473 Revert "Partial frame capture API part 6" by Ilya Nikolaevskiy · 6 years ago
  7. 85fc325 Revert "Partial frame capture API part 5" by Ilya Nikolaevskiy · 6 years ago
  8. 02f4e32 Make some new rtc_base targets publicly visible by Karl Wiberg · 6 years ago
  9. f13c2cd Roll chromium_revision eb2aa6ea6a..339f6a582b (630596:630696) by chromium-webrtc-autoroll · 6 years ago
  10. 61b4f74 Fix PeerConnectionInterface::StartRtcEventLog documentation. by Mirko Bonadei · 6 years ago
  11. 1a1c52b H.264 temporal layers w/frame marking (PART 2/3) by Johnny Lee · 6 years ago
  12. e556768 Roll chromium_revision eead273f0c..eb2aa6ea6a (630484:630596) by chromium-webrtc-autoroll · 6 years ago
  13. 157540a Stop hard-coding default IDs for RTP extensions by Elad Alon · 6 years ago
  14. efc9a14 Make UniqueNumberGenerator::AddKnownId() return a value by Elad Alon · 6 years ago
  15. 6ba2738 Roll chromium_revision d60317bbda..eead273f0c (630357:630484) by chromium-webrtc-autoroll · 6 years ago
  16. 5699142 Use c=IN IP4 <hostname> to support the presence of hostname candidates. by Qingsi Wang · 6 years ago
  17. 7832343 Revert "Enabling Simulcast use via AddTransceiver." by Emircan Uysaler · 6 years ago
  18. 836fee1 Calculate next process time in simulated network. by Sebastian Jansson · 6 years ago
  19. f6adac8 Add rtc event generic packet sent and received. by Piotr (Peter) Slatala · 6 years ago
  20. 50930a6 Roll chromium_revision 46a21d8d05..d60317bbda (630250:630357) by chromium-webrtc-autoroll · 6 years ago
  21. 1d13b37 Update LibvpxVp8Encoder to use EncodedImage::Allocate by Niels Möller · 6 years ago
  22. b7edf69 Delete rtc::File, usage replaced with FileWrapper by Niels Möller · 6 years ago
  23. 9f3aabb Delete obsolete class cricket::VideoCapturer by Niels Möller · 6 years ago
  24. 494ff28 Delete unused media constraints by Niels Möller · 6 years ago
  25. a8d48ab Fix incorrect FPS measure when frame dropper kicks in by Erik Språng · 6 years ago
  26. bdfadd6 Adds Stop methods to media streams in scenario framework. by Sebastian Jansson · 6 years ago
  27. 85eab49 Simplify peer connection smoke test to remove flakiness for now. by Artem Titov · 6 years ago
  28. 3dd473b Refactor of RtpPacket constructor by Johannes Kron · 6 years ago
  29. 7ff164e Plumbing of feedback on request setting by Johannes Kron · 6 years ago
  30. 5f6abcf Fix for RttBackoff when sending of packets with TWCC stops. by Christoffer Rodbro · 6 years ago
  31. dcba72b Resume rolling buildtools, now as chromium/src/buildtools by Oleh Prypin · 6 years ago
  32. b76b9ba Set WEBRTC_USE_H264 in common_config by Johannes Kron · 6 years ago
  33. 3f171df Add support for building iOS simulator code for iOS 11 and 12 by Artem Titarenko · 6 years ago
  34. 52e9e8d Remove now-unused iOS CI config files by Oleh Prypin · 6 years ago
  35. 51aa82d Roll chromium_revision 6f2fb1192a..46a21d8d05 (630145:630250) by chromium-webrtc-autoroll · 6 years ago
  36. 9f97c9a Add starting of VideoQualityAnalyzer in the e2e peer connection level test by Artem Titov · 6 years ago
  37. 5963fdd Pass-by-reference instead of value to initWithNativeEncodedImage by Dillon Cower · 6 years ago
  38. 108f20f Fix color space bug in wrapper of H264 decoder by Johannes Kron · 6 years ago
  39. a8cb366 Add field trial for forced software decoder fallback. by Åsa Persson · 6 years ago
  40. 587c5d1 Roll chromium_revision 34f99c21a3..6f2fb1192a (630023:630145) by chromium-webrtc-autoroll · 6 years ago
  41. ec3b9ff Move audio-related MediaTransport interfaces to their own file and target by Niels Möller · 6 years ago
  42. e12778c Update VP9EncoderImpl to use EncodedImage::Allocate by Niels Möller · 6 years ago
  43. f9a5561 Roll chromium_revision ee5dfb2215..34f99c21a3 (629907:630023) by chromium-webrtc-autoroll · 6 years ago
  44. d7180cc Also check the pending remote description when generating MIDs for legacy remote offers by Steve Anton · 6 years ago
  45. ce470aa Enabling Simulcast use via AddTransceiver. by Amit Hilbuch · 6 years ago
  46. a6a273d Introduce PeerConnectionE2EQualityTestFixture implementation. by Artem Titov · 6 years ago
  47. c363a53 Define RtpGenericFrameDescriptorExtension00 by Elad Alon · 6 years ago
  48. 260a71d Delete deprecated method PeerConnectionFactory::CreateVideoSource by Niels Möller · 6 years ago
  49. 59ab1cf Move ownership of RTPSenderVideo and RTPSenderAudio one level up by Niels Möller · 6 years ago
  50. 938dd9f Add owned data buffer to EncodedImage by Niels Möller · 6 years ago
  51. e6f6a0c Add missing operator= and extra methods to the SamplesStatsCounter. by Artem Titov · 6 years ago
  52. 710f3d3 Use task queue factory factory as parameter for TaskQueueTest by Danil Chapovalov · 6 years ago
  53. 0041fe5 Roll chromium_revision 1a597bc4e4..ee5dfb2215 (629788:629907) by chromium-webrtc-autoroll · 6 years ago
  54. cdab13d Roll chromium_revision c27b32b2fd..1a597bc4e4 (629510:629788) by Oleh Prypin · 6 years ago
  55. 86c8ad9 Pause rolling buildtools by Oleh Prypin · 6 years ago
  56. ef288dd Reland: Remove dead code from stream_params.h by Steve Anton · 6 years ago
  57. e1dcce2 Remove HAVE_WEBRTC_VOICE. by Fredrik Solenberg · 6 years ago
  58. e7b9e6b Move RtpSenderVideo tests to separate file. by Niels Möller · 6 years ago
  59. d70a114 Delete MediaTransport method SetNetworkChangeCallback by Niels Möller · 6 years ago
  60. fe6e50f Allow more than one registered network change callback in MediaTransport by Niels Möller · 6 years ago
  61. 3e61888 Roll chromium_revision 9d5d0c6635..c27b32b2fd (629245:629510) by Oleh Prypin · 6 years ago
  62. 7ca375c Implement encoder overshoot detector and rate adjuster. by Erik Språng · 6 years ago
  63. e98954c Prevent updating state in the delay manager if the packet was reordered. by Jakob Ivarsson · 6 years ago
  64. 9025bd5 Separate AndroidVideoTrackSource::OnFrameCaptured from adaptation by Magnus Jedvert · 6 years ago
  65. bb87f8a Delete unused/unsupported RetransmissionMode constants by Niels Möller · 6 years ago
  66. 0859142 Add events processing to GetIceEvents. by Sebastian Jansson · 6 years ago
  67. 4092d6f Fix autoroller to skip entries without @revision in them by Oleh Prypin · 6 years ago
  68. 6cfb403 Fix test FrameGenerator to work with a single file source by Ilya Nikolaevskiy · 6 years ago
  69. cf416e4 Revert "Remove dead code from stream_params.h" by Oleh Prypin · 6 years ago
  70. 2fb7999 Replace implicit int->char->string conversion by Oleh Prypin · 6 years ago
  71. 57d4ac9 Add more unit tests for RateControlSettings. by Rasmus Brandt · 6 years ago
  72. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  73. 9ce800d Add PRESUBMIT to enforce usage of new Googletest APIs. by Mirko Bonadei · 6 years ago
  74. 12d1285 Use the new TEST_SUITE GoogleTest API (regression). by Mirko Bonadei · 6 years ago
  75. 38c83b9 Remove unused file. by Fredrik Solenberg · 6 years ago
  76. 3f408d0 Remove dead code from stream_params.h by Steve Anton · 6 years ago
  77. d1b6206 Roll chromium_revision 3b81a4d714..9d5d0c6635 (629131:629245) by chromium-webrtc-autoroll · 6 years ago
  78. 65835be Allow logging of char* null pointer. by Niels Möller · 6 years ago
  79. 99b275d Introduce class that handles native wrapping of AndroidVideoTrackSource by Magnus Jedvert · 6 years ago
  80. b3032b6 Revert "Partial frame capture API part 4" by Ilya Nikolaevskiy · 6 years ago
  81. 7752ad6 Partial frame capture API part 6 by Ilya Nikolaevskiy · 6 years ago
  82. 1c54605 [clang-tidy] Apply performance-move-const-arg fixes (misc). by Mirko Bonadei · 6 years ago
  83. 87ce874 Allow link-time injection of the DefaultTaskQueueFactory by Danil Chapovalov · 6 years ago
  84. 93734c3 Roll chromium_revision b4fb8097f2..3b81a4d714 (628538:629131) by chromium-webrtc-autoroll · 6 years ago
  85. 9387b52 Apply simulcast resolution normalization before scaling. by Rasmus Brandt · 6 years ago
  86. 1f0a84a Partial frame capture API part 5 by Ilya Nikolaevskiy · 6 years ago
  87. 62b9fb4 Partial frame capture API part 4 by Ilya Nikolaevskiy · 6 years ago
  88. 9bee67c Add get/set base min delay to neteq and acm_receiver. by Ruslan Burakov · 6 years ago
  89. 9f6a0d5 In VideoEngine also respect requested TL number even for screenshare by Ilya Nikolaevskiy · 6 years ago
  90. b769894 Remove rule that discourages passing optional by const reference by Danil Chapovalov · 6 years ago
  91. 681de20 Stop changing the requested max bitrate based on protection level. by Rasmus Brandt · 6 years ago
  92. 167316b Remove proxy layer from AndroidVideoTrackSource by Magnus Jedvert · 6 years ago
  93. 69b761e Sets start on activities added after starting scenario test. by Sebastian Jansson · 6 years ago
  94. 30b182a New methods for registering network change callbacks in MediaTransport by Niels Möller · 6 years ago
  95. 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
  96. e22498c Compare GetStreamCaps against S_OK by Dan Minor · 6 years ago
  97. bfc9911 Remove TCPPort incoming_only_ member. by Mirko Bonadei · 6 years ago
  98. 167497f Delete VCMNackFecTable; appears unused for a long time. by Niels Möller · 6 years ago
  99. edbea46 Allow to change base minimum delay on NetEq. by Ruslan Burakov · 6 years ago
  100. d8b9804 Add scaleResolutionDownBy to RtpParameters.Encoding in Android SDK. by Mirta Dvornicic · 6 years ago