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gerrit-public.fairphone.software / platform / external / webrtc / 15f2ff46102741738683298f6d60158fa55b67a1
  1. 15f2ff4 VideoRtpReceiver: factor out VideoRtpTrackSource. by Markus Handell · 5 years ago
  2. 4995f87 Cleans up the round robin packet queue. by Erik Språng · 5 years ago
  3. 0174ffe Can list UWP applications by Julien Isorce · 5 years ago
  4. b0df593 Reland "Prepares PacingController for simplified packet queue." by Erik Språng · 5 years ago
  5. 6e9d0d3 Make base classes for native video encoder/decoder public by Xavier Lepaul · 5 years ago
  6. 58a3210 Add config to reduce weight on small samples in BitrateEstimator. by Christoffer Rodbro · 5 years ago
  7. 63737a9 Add new GOOG_PING and GOOG_MESSAGE_INTEGRITY_32 by Jonas Oreland · 5 years ago
  8. 026f64f Revert "VideoFrame: Store a reference to an encoded frame" by Markus Handell · 5 years ago
  9. 038fd99 Add RtpDepacketizerAv1::AssembleFrame function by Danil Chapovalov · 5 years ago
  10. ad020f5 Delete DEPRECATED_SingleThreadedTaskQueueForTesting by Danil Chapovalov · 5 years ago
  11. 76a7e51 [UBSan] Add suppressions for vp8, opus, absl. by Yves Gerey · 5 years ago
  12. 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 5 years ago
  13. ac7fd87 Force alignment of generated JVM called functions. by Raman Budny · 5 years ago
  14. ec22183 Revert lock for logging to CriticalSection by Danil Chapovalov · 5 years ago
  15. 9982efa MediaStreamInterface: introduce encoded sinks. by Markus Handell · 5 years ago
  16. b86a177 Expose ABGRToI420 in YuvHelper. by Sami Kalliomäki · 5 years ago
  17. e835fc0 Add UMA counter for audio interruptions by Henrik Lundin · 5 years ago
  18. c4e506e Fix writing into closed FrameWriter when dumping video in PC framework by Artem Titov · 5 years ago
  19. 22ae169 Remove dependencies on Chromium's webrtc_overrides. by Mirko Bonadei · 5 years ago
  20. 108a2f0 Preventively fix missing braces warnings. by Mirko Bonadei · 5 years ago
  21. 09c452e Split P2PTransportChannel by Jonas Oreland · 5 years ago
  22. d51cc7b Add absolute capture time property to rtp sources. by Ruslan Burakov · 5 years ago
  23. bb55e0b Clarifies identification of default communication device in ADM2 by henrika · 5 years ago
  24. eec1d04 Roll chromium_revision 8b5b046f02..022da7ca60 (715044:716991) by chromium-webrtc-autoroll · 5 years ago
  25. 6dd488b RTC_EXPORT RTCStatsMember's specialized members. by Mirko Bonadei · 5 years ago
  26. dca1449 Makes RoundRobinPacketQueue use same field trials as PacingController. by Erik Språng · 5 years ago
  27. 5831dda Introduce IVF file reader by Artem Titov · 5 years ago
  28. 1721de1 Add STUN_ATTR_GOOG_MISC_INFO by Jonas Oreland · 5 years ago
  29. 2f385d2 Manual chromium roll: Compile using JDK 11 by Yves Gerey · 5 years ago
  30. 3b1a8bb Account for rounding errors in dyanmic pacing mode. by Erik Språng · 5 years ago
  31. 83b2862 Protect against NumberOfEnumeratedDevices and Get[In|Out]putDeviceNames returning inconsistent results. by Tim Haloun · 5 years ago
  32. 4dd56a3 ACM: Adding unittests for the remixing functionality by Per Åhgren · 5 years ago
  33. 0e3198e Refactoring of the analog AGC functionality to add multichannel support by Per Åhgren · 5 years ago
  34. f3fcde3 Store delay measurements as struct instead of std::pair by Björn Terelius · 5 years ago
  35. fa7a8ca Revert "Prepares PacingController for simplified packet queue." by Erik Språng · 5 years ago
  36. 0660cee Add scale and compare methods to VideoFrame::UpdateRect by Ilya Nikolaevskiy · 5 years ago
  37. 332274d Adding GetInDtx to WebRTC Opus Interface. by Minyue Li · 5 years ago
  38. 67e5bd3 [Android SDK] Update to version 'Q' (29) to unblock autoroller/roll_deps.py by Yves Gerey · 5 years ago
  39. 063c7d1 In dependency descriptor remove extended fields indicator by Danil Chapovalov · 5 years ago
  40. fe04775 Fix a bug in interruption metrics by Henrik Lundin · 5 years ago
  41. 29e07e5 Add @Nullable annotations to quiet errorprone. by Yves Gerey · 5 years ago
  42. 0197887 Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester by Danil Chapovalov · 5 years ago
  43. 3574d05 EventLogVisualizer pacer delay plot fix. by philipel · 5 years ago
  44. 287e464 Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants by Rasmus Brandt · 5 years ago
  45. 8fe22fa StreamSynchronizationTest: Replace class Time with SimulatedClock. by Åsa Persson · 5 years ago
  46. db82cd7 Make android_compile_x64_rel actually compile in x64. by Yves Gerey · 5 years ago
  47. 351173c Tests that all available audio devices can be selected and used by the ADM. by henrika · 5 years ago
  48. 8ae70f6 Enable WebRTC-Bwe-MaxRttLimit by default. by Nikita Zetilov · 5 years ago
  49. 56d9452 Move stun.h to api/. by Patrik Höglund · 5 years ago
  50. aa3f5da Fork VCMPacket for PacketBuffer into own struct by Danil Chapovalov · 5 years ago
  51. acdc22d Prepares PacingController for simplified packet queue. by Erik Språng · 5 years ago
  52. ccf12c6 Reland "Add AV1 RtpDepacketizer class" by Danil Chapovalov · 5 years ago
  53. cb0b874 Add source-side perf upload script for WebRTC. by Patrik Höglund · 5 years ago
  54. 4186603 Always record timestamp of keyframe request. by Markus Handell · 5 years ago
  55. 92dd35d Reland "Force Chromium deps on the WebRTC component." by Mirko Bonadei · 5 years ago
  56. 9f99175 Revert "Add AV1 RtpDepacketizer class" by Yves Gerey · 5 years ago
  57. 49470c2 Add AV1 RtpDepacketizer class by Danil Chapovalov · 5 years ago
  58. 2419dc9 Revert "Force Chromium deps on the WebRTC component." by Mirko Bonadei · 5 years ago
  59. 313a10e p2p_transport_channel_unittest : put frequently used checks into functions. by Jonas Oreland · 5 years ago
  60. 39bab5a Add missing assert.h for win no-test build by Jerome Humbert · 5 years ago
  61. 408cb4b Make SCTPtransport enter "closed" state when DTLStransport does. by Harald Alvestrand · 5 years ago
  62. e6eded3 VideoFrame: Store a reference to an encoded frame by Markus Handell · 5 years ago
  63. ef6fe0c Use GetDefaultAudioEndpoint for the default communications device as well as the vanilla default device by Tim Haloun · 5 years ago
  64. 25ec888 Make ICE transports injectable. by Qingsi Wang · 5 years ago
  65. 53e157d Force Chromium deps on the WebRTC component. by Mirko Bonadei · 5 years ago
  66. bb55c5e Correct the upmixing of mono to stereo in ACM2 by Per Åhgren · 5 years ago
  67. fd0e32a Fix filtering of small packets in delay-based BWE by Björn Terelius · 5 years ago
  68. 7b46e17 In rtc::ByteBuffer drop support for ORDER_HOST as unused by Danil Chapovalov · 5 years ago
  69. 74f35e4 Add support for dynamic processing mode in PacedSender. by Erik Språng · 5 years ago
  70. 02fac7d Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  71. eb48799 Prepares PacingController for scheduled send tasks. by Erik Språng · 5 years ago
  72. 3527a4f In PacketBuffer split logic for detecting frame boundaries and assembling frame. by Danil Chapovalov · 5 years ago
  73. 01294f0 Don't configure video codec switching if no video stream has been created. by philipel · 5 years ago
  74. bb0aac2 Reduce verbosity of logging around playout underrun count on iOS. by Noah Richards · 5 years ago
  75. f11b46b Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."" by Mirko Bonadei · 5 years ago
  76. f181137 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  77. 1b66890 Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in call tests by Danil Chapovalov · 5 years ago
  78. 60bd1ae Detach thread checker in RtcEventLogImpl destructor. by Sebastian Jansson · 5 years ago
  79. 048b10a Correcting the ACM upmixing from mono/stereo to surround by Per Åhgren · 5 years ago
  80. 50f8686 Providing EmulatedRoute instances when creating TcpMessageRoute by Sebastian Jansson · 5 years ago
  81. 017c84f Synchronize is_screencast_ state in AndroidVideoTrackSource. by Jakob Ivarsson · 5 years ago
  82. c20a19c Allow extracting the linear AEC output by Per Åhgren · 5 years ago
  83. 06a394e Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."" by Mirko Bonadei · 5 years ago
  84. 57d53cf Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  85. 759f161 Use explicit template decl/def for RTCNonStandardStatsMember. by Mirko Bonadei · 5 years ago
  86. c5ec54e Add SetIsScreencast method to VideoSource. by Jakob Ivarsson · 5 years ago
  87. c1dac38 Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."" by Mirko Bonadei · 5 years ago
  88. f3dccc6 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  89. 1242d9c Reland Cleanup MultiStreamTester by Danil Chapovalov · 5 years ago
  90. d4002a7 RTC_EXPORT missing symbols for Chromium's component build. by Mirko Bonadei · 5 years ago
  91. f47c2ab RTC_EXPORT webrtc::RTCNonStandardStatsMember<T>. by Mirko Bonadei · 5 years ago
  92. 4ed17ff Roll chromium_revision 3b0967fe5b..0717a4dc31 (714463:714607) by chromium-webrtc-autoroll · 5 years ago
  93. 86b33e0 Don't ask for the friendly name of a default device if we failed to enumerate it. by Tim Haloun · 5 years ago
  94. dcc910a Adds simulated TCP message route for testing. by Sebastian Jansson · 5 years ago
  95. b4463ee Add ability to specify custom video source for PC framework. by Artem Titov · 5 years ago
  96. 815e00c Revert "Reset RtpFrameReferenceFinder on long pause" by Ilya Nikolaevskiy · 5 years ago
  97. 059daa4 Don't leak device moniker when BindToStorage fails. by Tim Haloun · 5 years ago
  98. 2bd2d85 Roll chromium_revision 3861228399..3b0967fe5b (714324:714463) by chromium-webrtc-autoroll · 5 years ago
  99. fe31b79 Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."" by Mirko Bonadei · 5 years ago
  100. 93f7907 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago