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gerrit-public.fairphone.software / platform / external / webrtc / 0da25a1c8e5de3b2f3c9e83280c96ae1d2ccb408
  1. 0da25a1 Update TransportSequenceNumberV2 extension to support fixed size by Johannes Kron · 5 years ago
  2. f441ea9 Minor cleanup of probe_controller.cc by Jonas Olsson · 5 years ago
  3. 200feba Make AEC3 the default desktop AEC option in WebRTC by Per Åhgren · 5 years ago
  4. 359c9b9 Roll chromium_revision 49f30ad2d0..99baeeafe2 (637641:638035) by chromium-webrtc-autoroll · 5 years ago
  5. be7af93 Add dsymutil as a mac cipd dependency. by Yves Gerey · 5 years ago
  6. b443dfe Use metrics::Samples in a couple pc/ tests by Steve Anton · 5 years ago
  7. e2a284d Adding metrics to measure usage of simulcast API. by Amit Hilbuch · 5 years ago
  8. 4eb5c14 Reland "Remove field trial include from decision logic." by Jakob Ivarsson · 5 years ago
  9. 07a4f2b Merge rtc_task_queue(_api|_impl)? build targets into one by Danil Chapovalov · 5 years ago
  10. 4450708 Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics. by Jakob Ivarsson · 5 years ago
  11. 1aa7581 Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask by Danil Chapovalov · 5 years ago
  12. c1e6e86 Add metrics::Samples to facilitate easier testing by Steve Anton · 5 years ago
  13. d36c086 Add support for simulcast streams in QualityAnalyzingVideoDecoder. by Artem Titov · 5 years ago
  14. 6ec2f54 Fix mis-spelled TODO items by Niels Möller · 5 years ago
  15. 7949f21 Revert "Removes lock from ChannelSend." by Sebastian Jansson · 5 years ago
  16. 9ef5e05 Fix target bitrate handling for a single layer VP9 screenshare by Ilya Nikolaevskiy · 5 years ago
  17. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 5 years ago
  18. f23f161 Roll chromium_revision 5fe3fd14f6..49f30ad2d0 (637538:637641) by chromium-webrtc-autoroll · 5 years ago
  19. 423bae4 Revert "Remove field trial include from decision logic." by Jakob Ivarsson‎ · 5 years ago
  20. bd7ed4b Include sign for infinity in ToString for data units. by Sebastian Jansson · 5 years ago
  21. 44ce4b4 Adding a placeholder audio_buffer build target by Per Åhgren · 5 years ago
  22. 15845af Reland "Another mock for GetSctpTransport" (and add test) by Harald Alvestrand · 5 years ago
  23. 9b93447 Removes lock from ChannelSend. by Sebastian Jansson · 5 years ago
  24. c8eeb18 Fixes parsing of logs where receive time info is missing. by Sebastian Jansson · 5 years ago
  25. 07316a6 Roll chromium_revision 4229a4b64d..5fe3fd14f6 (637436:637538) by chromium-webrtc-autoroll · 5 years ago
  26. 1f8e445 Roll chromium_revision 5afa522447..4229a4b64d (637301:637436) by chromium-webrtc-autoroll · 5 years ago
  27. 572c60f Injecting Clock into video senders. by Sebastian Jansson · 5 years ago
  28. 8026d60 Injecting Clock in video receive. by Sebastian Jansson · 5 years ago
  29. ef50b25 Remove lock in WebRtcVideoEngine by Steve Anton · 5 years ago
  30. 4cde9ad Fix some typos found in ivf_file_writer.cc by Elad Alon · 5 years ago
  31. 4e5f5ed Allow Clock injection in Call. by Sebastian Jansson · 5 years ago
  32. 5fe9510 Move ownership of RTPSenderVideo one more level up, to RtpVideoSender by Niels Möller · 5 years ago
  33. ac6cf7f Roll chromium_revision e65d7afd91..5afa522447 (637200:637301) by chromium-webrtc-autoroll · 5 years ago
  34. da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 5 years ago
  35. d0f3d84 Wire UpdateRect signalling in test frame generators by Ilya Nikolaevskiy · 5 years ago
  36. acd8ae7 Reinstate old iceConnectionState "completed" behavior by Jonas Olsson · 5 years ago
  37. 0a16916 Use JavaAudioDeviceModule as default by Paulina Hensman · 5 years ago
  38. 13471a4 Switch back to native mutexes on macOS by Oskar Sundbom · 5 years ago
  39. b678940 Using send time instead of system clock in quality scaler. by Sebastian Jansson · 5 years ago
  40. e64a688 Replacing Clock in ScreenshareLayers. by Sebastian Jansson · 5 years ago
  41. c130d42 Add ability to unwind stack for the current thread by Karl Wiberg · 5 years ago
  42. 8b8d01a Add full stack test with weak 3g-like properties by Erik Språng · 5 years ago
  43. 727504c Revert "Another mock for GetSctpTransport" by Harald Alvestrand · 5 years ago
  44. 3b548dd Move rtc::NewClosure into own build target as ToQueuedTask by Danil Chapovalov · 5 years ago
  45. b2c4700 Another mock for GetSctpTransport by Harald Alvestrand · 5 years ago
  46. 87e05b5 NetEq fuzzer: lower the maximum fuzzer input size by Henrik Lundin · 5 years ago
  47. 7ceef35 Roll chromium_revision b3ef4b21cb..e65d7afd91 (637096:637200) by chromium-webrtc-autoroll · 5 years ago
  48. 4a42742 Make rtc_base/fake_mdns_responder.h self contained. by Mirko Bonadei · 5 years ago
  49. 1916cbc Fix -Winconsistent-missing-override in fake_network.h. by Mirko Bonadei · 5 years ago
  50. c9ea545 Roll chromium_revision 1951ee5099..b3ef4b21cb (636995:637096) by chromium-webrtc-autoroll · 5 years ago
  51. 7e215c6 Roll chromium_revision ac8660421f..1951ee5099 (636869:636995) by chromium-webrtc-autoroll · 5 years ago
  52. aabd036 Simulcast should be disabled if RID header extension is not supported. by Amit Hilbuch · 5 years ago
  53. b1ae10b Add x-mt line to the offer. by Piotr (Peter) Slatala · 5 years ago
  54. 896b47c Injecting ProcessThread and TaskQueueFactory in Call. by Sebastian Jansson · 5 years ago
  55. 52426ed Modify BufferedFrameDecryptor to perform fine grained key requests. by Benjamin Wright · 5 years ago
  56. e4bd9a1 Style guide fixes for the hkdf class. by Benjamin Wright · 5 years ago
  57. baffae6 Roll chromium_revision 8eb8e09f19..ac8660421f (636762:636869) by chromium-webrtc-autoroll · 5 years ago
  58. ed50e6c Inject TaskQueueFactory in RtpTransportControllerSend. by Sebastian Jansson · 5 years ago
  59. 4765013 Intermediate step: Move ownership of rtc::NetworkManager to test code from PC E2E framework by Artem Titov · 5 years ago
  60. 547a1dc Removes injection of RtpTransportControllerSend from Call::Create. by Sebastian Jansson · 5 years ago
  61. d9f798a Remove field trial include from decision logic. by Jakob Ivarsson · 5 years ago
  62. d1d0359 Remove memsets of CodecSpecificInfo. by philipel · 5 years ago
  63. 2997ec9 Removes unused keep-alive from RtpTransportControllerSend. by Sebastian Jansson · 5 years ago
  64. 8452a9e Roll chromium_revision 24eaf090c6..8eb8e09f19 (636660:636762) by chromium-webrtc-autoroll · 5 years ago
  65. 74682c1 Inject TaskQueueFactory to video streams. by Sebastian Jansson · 5 years ago
  66. 859abef Use DefaultVideoQualityAnalyzer as default, cleanup headers. by Artem Titov · 5 years ago
  67. c68ddd1 Fix namespace for PeerConnectionE2EQualityTestFixture by Artem Titov · 5 years ago
  68. fc52b91 Implicitly suppress //build/config/clang:find_bad_constructs. by Mirko Bonadei · 5 years ago
  69. 3830d9b Fix peerconnection_quality_test #includes and deps. by Mirko Bonadei · 5 years ago
  70. 328027b Replace fatal error with error log by Danil Chapovalov · 5 years ago
  71. cdea67d Add GetSctpTransport to proxy map by Harald Alvestrand · 5 years ago
  72. 6fe413d sdk/android:native_api_stacktrace: Declare a more narrow set of dependencies by Karl Wiberg · 5 years ago
  73. 06c31f6 Roll chromium_revision d1e2a1cf94..24eaf090c6 (636518:636660) by chromium-webrtc-autoroll · 5 years ago
  74. 8e98c60 Cleanup for openssl_stream_adapter.cc. by Benjamin Wright · 5 years ago
  75. df5923d scale_resolution_down_by and rid are implemented by Steve Anton · 5 years ago
  76. 9ded485 Implement OpenChannel() on test media transports and make it pure virtual. by Bjorn Mellem · 5 years ago
  77. 766f624 Roll chromium_revision 16b0680682..d1e2a1cf94 (636404:636518) by chromium-webrtc-autoroll · 5 years ago
  78. 9a7e721 Use default values for video and audio streams generation in PC E2E framework by Artem Titov · 5 years ago
  79. fb14c5d Allow injection of TaskQueueFactory in FrameGeneratorCapturer. by Sebastian Jansson · 5 years ago
  80. b8a4d68 Allow injection of task queue factory in RtcEventLog. by Sebastian Jansson · 5 years ago
  81. 8ea0238 Add bandwidth floor for RTT based backoff. by Christoffer Rodbro · 5 years ago
  82. 3cdd4d5 Fix: Ignore empty frames in Media Transport by Piotr (Peter) Slatala · 5 years ago
  83. 26c59ff Fix jitter buffer delay reporting. by Jakob Ivarsson · 5 years ago
  84. c58c01d Add construtor from required fields for VideoConfig in PC E2E framework by Artem Titov · 5 years ago
  85. 3481db2 Add stream label to audio streams in PC E2E framework by Artem Titov · 5 years ago
  86. 970f2f7 [clang-tidy] Apply bugprone-argument-comment fixes. by Mirko Bonadei · 5 years ago
  87. d3a780b Cleanup NetEqPostponeDecodingAfterExpand field trial. by Jakob Ivarsson · 5 years ago
  88. 6b7bf6a Add a presubmit check for absl/memory/memory.h inclusion for WrapUnique by tzik · 5 years ago
  89. cf7c584 Only draw frames with height and width >0 by Paulina Hensman · 5 years ago
  90. 40b030e Reland "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Henrik Boström · 5 years ago
  91. 25fb765 Roll chromium_revision ec3bf6e607..16b0680682 (635450:636404) by chromium-webrtc-autoroll · 5 years ago
  92. ba4dcc3 rtc::Thread::PostTask() added. by Henrik Boström · 5 years ago
  93. 8f385e3 Remove dependency on DECLARE_IUNKNOWN macro on Windows. by Tommi · 5 years ago
  94. 7c55415 Fix call setup: change way of adding media to the call. by Artem Titov · 5 years ago
  95. 03257b0 Add flag for explicitly specifying that the legacy AEC2 should be used by Per Åhgren · 5 years ago
  96. f3280e9 Create conversions between webrtc::TaskQueueBase and rtc::TaskQueue by Danil Chapovalov · 5 years ago
  97. c85328f Add SCTP transport to the public API. by Harald Alvestrand · 5 years ago
  98. 60fd73a Migrate SequencedTaskChecker to rely on webrtc::TaskQueueBase interface by Danil Chapovalov · 5 years ago
  99. ba7886b Move command line flags out of NetEqTestFactory by Ivo Creusen · 5 years ago
  100. 5983585 Introduce test case name in peer connection e2e test framework. by Artem Titov · 5 years ago