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gerrit-public.fairphone.software / platform / external / webrtc / de3360ec1dd38c702415875a81e9fe3bb42cb935
  1. de3360e Create Vp8FrameBufferController by Elad Alon · 6 years ago
  2. 610c763 Add target bitrate headroom signal to VideoStreamEncoder. by Erik Språng · 6 years ago
  3. e49d64e Roll chromium_revision 3eb6e6ce76..1af146a0f6 (638159:638325) by chromium-webrtc-autoroll · 6 years ago
  4. 7276b97 Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC. by Benjamin Wright · 6 years ago
  5. 4423c36 Migrate RepeatingTask to take raw pointer to TaskQueueBase instead of TaskQueue by Danil Chapovalov · 6 years ago
  6. 11e55ee Renaming min_pacing_rate to min_total_allocated_bitrate. by Sebastian Jansson · 6 years ago
  7. 7b41225 Throttle frame-rate In VP8 encoder in steady state for screenshare by Ilya Nikolaevskiy · 6 years ago
  8. 2ecc8c8 Roll chromium_revision 99baeeafe2..3eb6e6ce76 (638035:638159) by chromium-webrtc-autoroll · 6 years ago
  9. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  10. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  11. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  12. 67f862e Guard against calls to OnEncodedFrame after Release. by Sami Kalliomäki · 6 years ago
  13. 6117068 Throttle frame-rate In VP9 encoder in steady state for screenshare by Ilya Nikolaevskiy · 6 years ago
  14. 0cb858c New VCMPacket constructor without WebRtcRTPHeader argument by Niels Möller · 6 years ago
  15. 7bc331f Android: Use android_deps directly by Peter Wen · 6 years ago
  16. c44f6cc Modernize RtpRtcp factory function: use unique_ptr as return type by Danil Chapovalov · 6 years ago
  17. ede7cb2 Rewrite video_loopback to use new mac capturer. by Kári Tristan Helgason · 6 years ago
  18. c8d2e73 Delete CodecSpecificInfo argument from VideoEncoder::Encode by Niels Möller · 6 years ago
  19. 1e42761 Removes verbose extension warning in Scenario tests. by Sebastian Jansson · 6 years ago
  20. 110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 6 years ago
  21. 745cfb9 use link_deps in ana_debug_dump_proto by Takuto Ikuta · 6 years ago
  22. d738071 Refactor FakeEncoder to avoid writing to a const EncodedImage by Niels Möller · 6 years ago
  23. 9df3353 Generic Frame Descriptor (GFD) VP8 templates. by philipel · 6 years ago
  24. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  25. 0da25a1 Update TransportSequenceNumberV2 extension to support fixed size by Johannes Kron · 6 years ago
  26. f441ea9 Minor cleanup of probe_controller.cc by Jonas Olsson · 6 years ago
  27. 200feba Make AEC3 the default desktop AEC option in WebRTC by Per Åhgren · 6 years ago
  28. 359c9b9 Roll chromium_revision 49f30ad2d0..99baeeafe2 (637641:638035) by chromium-webrtc-autoroll · 6 years ago
  29. be7af93 Add dsymutil as a mac cipd dependency. by Yves Gerey · 6 years ago
  30. b443dfe Use metrics::Samples in a couple pc/ tests by Steve Anton · 6 years ago
  31. e2a284d Adding metrics to measure usage of simulcast API. by Amit Hilbuch · 6 years ago
  32. 4eb5c14 Reland "Remove field trial include from decision logic." by Jakob Ivarsson · 6 years ago
  33. 07a4f2b Merge rtc_task_queue(_api|_impl)? build targets into one by Danil Chapovalov · 6 years ago
  34. 4450708 Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics. by Jakob Ivarsson · 6 years ago
  35. 1aa7581 Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask by Danil Chapovalov · 6 years ago
  36. c1e6e86 Add metrics::Samples to facilitate easier testing by Steve Anton · 6 years ago
  37. d36c086 Add support for simulcast streams in QualityAnalyzingVideoDecoder. by Artem Titov · 6 years ago
  38. 6ec2f54 Fix mis-spelled TODO items by Niels Möller · 6 years ago
  39. 7949f21 Revert "Removes lock from ChannelSend." by Sebastian Jansson · 6 years ago
  40. 9ef5e05 Fix target bitrate handling for a single layer VP9 screenshare by Ilya Nikolaevskiy · 6 years ago
  41. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  42. f23f161 Roll chromium_revision 5fe3fd14f6..49f30ad2d0 (637538:637641) by chromium-webrtc-autoroll · 6 years ago
  43. 423bae4 Revert "Remove field trial include from decision logic." by Jakob Ivarsson‎ · 6 years ago
  44. bd7ed4b Include sign for infinity in ToString for data units. by Sebastian Jansson · 6 years ago
  45. 44ce4b4 Adding a placeholder audio_buffer build target by Per Åhgren · 6 years ago
  46. 15845af Reland "Another mock for GetSctpTransport" (and add test) by Harald Alvestrand · 6 years ago
  47. 9b93447 Removes lock from ChannelSend. by Sebastian Jansson · 6 years ago
  48. c8eeb18 Fixes parsing of logs where receive time info is missing. by Sebastian Jansson · 6 years ago
  49. 07316a6 Roll chromium_revision 4229a4b64d..5fe3fd14f6 (637436:637538) by chromium-webrtc-autoroll · 6 years ago
  50. 1f8e445 Roll chromium_revision 5afa522447..4229a4b64d (637301:637436) by chromium-webrtc-autoroll · 6 years ago
  51. 572c60f Injecting Clock into video senders. by Sebastian Jansson · 6 years ago
  52. 8026d60 Injecting Clock in video receive. by Sebastian Jansson · 6 years ago
  53. ef50b25 Remove lock in WebRtcVideoEngine by Steve Anton · 6 years ago
  54. 4cde9ad Fix some typos found in ivf_file_writer.cc by Elad Alon · 6 years ago
  55. 4e5f5ed Allow Clock injection in Call. by Sebastian Jansson · 6 years ago
  56. 5fe9510 Move ownership of RTPSenderVideo one more level up, to RtpVideoSender by Niels Möller · 6 years ago
  57. ac6cf7f Roll chromium_revision e65d7afd91..5afa522447 (637200:637301) by chromium-webrtc-autoroll · 6 years ago
  58. da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
  59. d0f3d84 Wire UpdateRect signalling in test frame generators by Ilya Nikolaevskiy · 6 years ago
  60. acd8ae7 Reinstate old iceConnectionState "completed" behavior by Jonas Olsson · 6 years ago
  61. 0a16916 Use JavaAudioDeviceModule as default by Paulina Hensman · 6 years ago
  62. 13471a4 Switch back to native mutexes on macOS by Oskar Sundbom · 6 years ago
  63. b678940 Using send time instead of system clock in quality scaler. by Sebastian Jansson · 6 years ago
  64. e64a688 Replacing Clock in ScreenshareLayers. by Sebastian Jansson · 6 years ago
  65. c130d42 Add ability to unwind stack for the current thread by Karl Wiberg · 6 years ago
  66. 8b8d01a Add full stack test with weak 3g-like properties by Erik Språng · 6 years ago
  67. 727504c Revert "Another mock for GetSctpTransport" by Harald Alvestrand · 6 years ago
  68. 3b548dd Move rtc::NewClosure into own build target as ToQueuedTask by Danil Chapovalov · 6 years ago
  69. b2c4700 Another mock for GetSctpTransport by Harald Alvestrand · 6 years ago
  70. 87e05b5 NetEq fuzzer: lower the maximum fuzzer input size by Henrik Lundin · 6 years ago
  71. 7ceef35 Roll chromium_revision b3ef4b21cb..e65d7afd91 (637096:637200) by chromium-webrtc-autoroll · 6 years ago
  72. 4a42742 Make rtc_base/fake_mdns_responder.h self contained. by Mirko Bonadei · 6 years ago
  73. 1916cbc Fix -Winconsistent-missing-override in fake_network.h. by Mirko Bonadei · 6 years ago
  74. c9ea545 Roll chromium_revision 1951ee5099..b3ef4b21cb (636995:637096) by chromium-webrtc-autoroll · 6 years ago
  75. 7e215c6 Roll chromium_revision ac8660421f..1951ee5099 (636869:636995) by chromium-webrtc-autoroll · 6 years ago
  76. aabd036 Simulcast should be disabled if RID header extension is not supported. by Amit Hilbuch · 6 years ago
  77. b1ae10b Add x-mt line to the offer. by Piotr (Peter) Slatala · 6 years ago
  78. 896b47c Injecting ProcessThread and TaskQueueFactory in Call. by Sebastian Jansson · 6 years ago
  79. 52426ed Modify BufferedFrameDecryptor to perform fine grained key requests. by Benjamin Wright · 6 years ago
  80. e4bd9a1 Style guide fixes for the hkdf class. by Benjamin Wright · 6 years ago
  81. baffae6 Roll chromium_revision 8eb8e09f19..ac8660421f (636762:636869) by chromium-webrtc-autoroll · 6 years ago
  82. ed50e6c Inject TaskQueueFactory in RtpTransportControllerSend. by Sebastian Jansson · 6 years ago
  83. 4765013 Intermediate step: Move ownership of rtc::NetworkManager to test code from PC E2E framework by Artem Titov · 6 years ago
  84. 547a1dc Removes injection of RtpTransportControllerSend from Call::Create. by Sebastian Jansson · 6 years ago
  85. d9f798a Remove field trial include from decision logic. by Jakob Ivarsson · 6 years ago
  86. d1d0359 Remove memsets of CodecSpecificInfo. by philipel · 6 years ago
  87. 2997ec9 Removes unused keep-alive from RtpTransportControllerSend. by Sebastian Jansson · 6 years ago
  88. 8452a9e Roll chromium_revision 24eaf090c6..8eb8e09f19 (636660:636762) by chromium-webrtc-autoroll · 6 years ago
  89. 74682c1 Inject TaskQueueFactory to video streams. by Sebastian Jansson · 6 years ago
  90. 859abef Use DefaultVideoQualityAnalyzer as default, cleanup headers. by Artem Titov · 6 years ago
  91. c68ddd1 Fix namespace for PeerConnectionE2EQualityTestFixture by Artem Titov · 6 years ago
  92. fc52b91 Implicitly suppress //build/config/clang:find_bad_constructs. by Mirko Bonadei · 6 years ago
  93. 3830d9b Fix peerconnection_quality_test #includes and deps. by Mirko Bonadei · 6 years ago
  94. 328027b Replace fatal error with error log by Danil Chapovalov · 6 years ago
  95. cdea67d Add GetSctpTransport to proxy map by Harald Alvestrand · 6 years ago
  96. 6fe413d sdk/android:native_api_stacktrace: Declare a more narrow set of dependencies by Karl Wiberg · 6 years ago
  97. 06c31f6 Roll chromium_revision d1e2a1cf94..24eaf090c6 (636518:636660) by chromium-webrtc-autoroll · 6 years ago
  98. 8e98c60 Cleanup for openssl_stream_adapter.cc. by Benjamin Wright · 6 years ago
  99. df5923d scale_resolution_down_by and rid are implemented by Steve Anton · 6 years ago
  100. 9ded485 Implement OpenChannel() on test media transports and make it pure virtual. by Bjorn Mellem · 6 years ago