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gerrit-public.fairphone.software / platform / external / webrtc / 05d43c6f7fe497fed0f2c8714e2042dd07a86df2
  1. 05d43c6 Fix getStats() freeze bug affecting Chromium but not WebRTC standalone. by Henrik Boström · 6 years ago
  2. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  3. 106d92d Add thread safety annotation for PeerConnection::SignalDataChannelCreated_ by Karl Wiberg · 6 years ago
  4. 13bc871 PostMessageWithFunctor() added. by Henrik Boström · 6 years ago
  5. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  6. 7e0e44f Move video-related MediaTransport interfaces to their own file and target by Niels Möller · 6 years ago
  7. 054db54 Remove an absl::WrapUnique usage without absl/memory/memory.h include by tzik · 6 years ago
  8. 22997d6 Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477) by chromium-webrtc-autoroll · 6 years ago
  9. 1c9c9fc Replace replace_substrs with Abseil by Steve Anton · 6 years ago
  10. bf9e01a Add support of fast media sending in peer connection e2e test by Artem Titov · 6 years ago
  11. ceba6ae Return a copy, becase GetPercentile in SamplesStatsCounter isn't const by Artem Titov · 6 years ago
  12. cf8405e Add generic packet rates to event_log_visualizer. by Piotr (Peter) Slatala · 6 years ago
  13. 15653f9 Roll chromium_revision 78de17c053..2ad52fb2a4 (632252:632357) by chromium-webrtc-autoroll · 6 years ago
  14. aa58415 Reland "Enabling Simulcast use via AddTransceiver." by Amit Hilbuch · 6 years ago
  15. aec9794 Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format. by Piotr (Peter) Slatala · 6 years ago
  16. 9e2692c Roll chromium_revision 9a34b2cc2d..78de17c053 (632146:632252) by chromium-webrtc-autoroll · 6 years ago
  17. d036c65 Clarify and unify outgoing and incoming packet loss rate plots. by Konrad Hofbauer · 6 years ago
  18. 663844d Update test code to use EncodedImage::Allocate by Niels Möller · 6 years ago
  19. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  20. 92e7c69 Revert "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Moller · 6 years ago
  21. 8e847ee Make recv_deltas optional in TransportFeedback packets by Johannes Kron · 6 years ago
  22. 69fb6c8 Allow DtlsTransport::Information() to be called off-thread by Harald Alvestrand · 6 years ago
  23. 068fc35 Break out parameters from EventLogAnalyzer to AnalyzerConfig struct. by Bjorn Terelius · 6 years ago
  24. f0c366b Cleanup of scenario test video stream setup. by Sebastian Jansson · 6 years ago
  25. d00045e Changing command line flag for scenario logs root directory. by Sebastian Jansson · 6 years ago
  26. dac03d9 Move MediaConstraintsInterface to sdk/, and make it a concrete class by Niels Möller · 6 years ago
  27. 1d7bf89 Add LS_VERBOSE logging for target bitrate in GoogCC by Evan Shrubsole · 6 years ago
  28. 0179a3d Roll chromium_revision bfd7bcf815..9a34b2cc2d (632040:632146) by chromium-webrtc-autoroll · 6 years ago
  29. da825b1 Replace NOTREACHED with a break. by Piotr (Peter) Slatala · 6 years ago
  30. 1290fc7 Remove old accessor in GenericAckReceived by Piotr (Peter) Slatala · 6 years ago
  31. 788f577 Update the resolution check for VP8 simulcast. by Mirta Dvornicic · 6 years ago
  32. 6b88a8f Introduce default video quality analyzer by Artem Titov · 6 years ago
  33. b1ea48c Roll chromium_revision 103665932a..bfd7bcf815 (631883:632040) by chromium-webrtc-autoroll · 6 years ago
  34. 5ae259e Use a provider in rtc::Network to access the mDNS responder. by Qingsi Wang · 6 years ago
  35. 616b233 Add FullStackTest with simulated encoder overshooting by Erik Språng · 6 years ago
  36. 6c02541 Revert "Delete video source proxying in WebRtcVideoSendStream" by Christian Fremerey · 6 years ago
  37. 3588394 Roll chromium_revision d026ac796d..103665932a (631722:631883) by chromium-webrtc-autoroll · 6 years ago
  38. dfd5c4b Parse XR, FIR and PLI in rtc_event_log_parser.cc by Bjorn Terelius · 6 years ago
  39. 3c119fb Handle HKDF key derivation when building with OpenSSL. by Sergey Sablin · 6 years ago
  40. 5e2aad1 Support GenericPacketReceived/Sent/AckReceived event logs. by Piotr (Peter) Slatala · 6 years ago
  41. 975a899 Roll chromium_revision aa7b61fdc4..d026ac796d (631597:631722) by chromium-webrtc-autoroll · 6 years ago
  42. 4a68fb9 Separate base minimum delay and minimum delay. by Ruslan Burakov · 6 years ago
  43. 69bb3af Update EncodedFrameForMediaTransport to use Retain() rather than set_buffer + memcpy. by Niels Möller · 6 years ago
  44. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
  45. cbf5949 Update MultiplexEncoderAdapter to use EncodedImage::Allocate by Niels Möller · 6 years ago
  46. 448c387 IceTransportWithTransportChannel: Initialize |thread_checker_| in declaration by Raphael Kubo da Costa · 6 years ago
  47. 2bd54a1 Ensure TestPeers are destroyed at the end of Run. by Mirko Bonadei · 6 years ago
  48. 6aca0b7 Add |update_rect| field and UpdateRect struct to VideoFrame. by Ilya Nikolaevskiy · 6 years ago
  49. 7f24fb9 Add settings to turn off VP8 base layer qp limit by Erik Språng · 6 years ago
  50. 98bcd32 Remove always_passing_unittest.cc. by Mirko Bonadei · 6 years ago
  51. b4f7ab1 Fix -Wunused-result warnings by Hans Wennborg · 6 years ago
  52. eedb0a1 Roll chromium_revision 23b4d2134b..aa7b61fdc4 (631425:631597) by chromium-webrtc-autoroll · 6 years ago
  53. a795c3b Roll chromium_revision d366835eb8..23b4d2134b (631269:631425) by chromium-webrtc-autoroll · 6 years ago
  54. dcbdd2c Add Foundation.framework to cocoa_threading target by Jiawei Ou · 6 years ago
  55. 0874530 Add gn configs to remove the dependency to audio and video codecs. by Jiawei Ou · 6 years ago
  56. 3329be4 Roll chromium_revision b847e52039..d366835eb8 (631155:631269) by chromium-webrtc-autoroll · 6 years ago
  57. 464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
  58. eb81b47 Update H264EncoderImpl to use EncodedImage::Allocate by Niels Möller · 6 years ago
  59. d3666b2 Introduce cross traffic for emulated network layer. by Artem Titov · 6 years ago
  60. 5c4ddad Delete obsolete usage of FakeConstraints by Niels Möller · 6 years ago
  61. 9bf67ea AEC3: Fix delay hysteresis validation by Gustaf Ullberg · 6 years ago
  62. 99b9149 Enable padding bit in TransportFeedback packets by Johannes Kron · 6 years ago
  63. 2ce0cb0 Add missing 'explicit' specifier to GainControlImpl by Sam Zackrisson · 6 years ago
  64. eb17524 Migrate libevent task queue implementation to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  65. 675e5aa Roll chromium_revision b6a69427be..b847e52039 (631040:631155) by chromium-webrtc-autoroll · 6 years ago
  66. a93b8b0 Update SimulcastTestFixtureImpl to use EncodedImage::Allocate by Niels Möller · 6 years ago
  67. 40027b1 Roll chromium_revision e4a7c15e8a..b6a69427be (630925:631040) by chromium-webrtc-autoroll · 6 years ago
  68. a887c32 Roll chromium_revision 92e6bfe90b..e4a7c15e8a (630806:630925) by chromium-webrtc-autoroll · 6 years ago
  69. 26d0876 Roll chromium_revision 339f6a582b..92e6bfe90b (630696:630806) by chromium-webrtc-autoroll · 6 years ago
  70. 871e144 Revert "Reland "Partial frame capture API part 1"" by Ilya Nikolaevskiy · 6 years ago
  71. 421c859 Remove crit_render_ lock from webrtc::GainControlImpl by Sam Zackrisson · 6 years ago
  72. 00f9400 Dump histogram data in AEC3 delay estimator by Sam Zackrisson · 6 years ago
  73. 271195f Fix potential crash when building rtx packet by Danil Chapovalov · 6 years ago
  74. 501bfba Split rtp_receiver for readability. by Ruslan Burakov · 6 years ago
  75. b66003c Delete video source proxying in WebRtcVideoSendStream by Niels Möller · 6 years ago
  76. 6df89cc Revert "Partial frame capture API part 2" by Ilya Nikolaevskiy · 6 years ago
  77. b00eb19 Removes Start/Stop on network emulation manager. by Sebastian Jansson · 6 years ago
  78. eb7589e Revert "Partial frame capture API part 3" by Ilya Nikolaevskiy · 6 years ago
  79. fd5d473 Revert "Partial frame capture API part 6" by Ilya Nikolaevskiy · 6 years ago
  80. 85fc325 Revert "Partial frame capture API part 5" by Ilya Nikolaevskiy · 6 years ago
  81. 02f4e32 Make some new rtc_base targets publicly visible by Karl Wiberg · 6 years ago
  82. f13c2cd Roll chromium_revision eb2aa6ea6a..339f6a582b (630596:630696) by chromium-webrtc-autoroll · 6 years ago
  83. 61b4f74 Fix PeerConnectionInterface::StartRtcEventLog documentation. by Mirko Bonadei · 6 years ago
  84. 1a1c52b H.264 temporal layers w/frame marking (PART 2/3) by Johnny Lee · 6 years ago
  85. e556768 Roll chromium_revision eead273f0c..eb2aa6ea6a (630484:630596) by chromium-webrtc-autoroll · 6 years ago
  86. 157540a Stop hard-coding default IDs for RTP extensions by Elad Alon · 6 years ago
  87. efc9a14 Make UniqueNumberGenerator::AddKnownId() return a value by Elad Alon · 6 years ago
  88. 6ba2738 Roll chromium_revision d60317bbda..eead273f0c (630357:630484) by chromium-webrtc-autoroll · 6 years ago
  89. 5699142 Use c=IN IP4 <hostname> to support the presence of hostname candidates. by Qingsi Wang · 6 years ago
  90. 7832343 Revert "Enabling Simulcast use via AddTransceiver." by Emircan Uysaler · 6 years ago
  91. 836fee1 Calculate next process time in simulated network. by Sebastian Jansson · 6 years ago
  92. f6adac8 Add rtc event generic packet sent and received. by Piotr (Peter) Slatala · 6 years ago
  93. 50930a6 Roll chromium_revision 46a21d8d05..d60317bbda (630250:630357) by chromium-webrtc-autoroll · 6 years ago
  94. 1d13b37 Update LibvpxVp8Encoder to use EncodedImage::Allocate by Niels Möller · 6 years ago
  95. b7edf69 Delete rtc::File, usage replaced with FileWrapper by Niels Möller · 6 years ago
  96. 9f3aabb Delete obsolete class cricket::VideoCapturer by Niels Möller · 6 years ago
  97. 494ff28 Delete unused media constraints by Niels Möller · 6 years ago
  98. a8d48ab Fix incorrect FPS measure when frame dropper kicks in by Erik Språng · 6 years ago
  99. bdfadd6 Adds Stop methods to media streams in scenario framework. by Sebastian Jansson · 6 years ago
  100. 85eab49 Simplify peer connection smoke test to remove flakiness for now. by Artem Titov · 6 years ago