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gerrit-public.fairphone.software / platform / external / webrtc / e01857cca48f8838c3e7d11a67ca377a3e355c60
  1. e01857c Revert "Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"" by Ilya Nikolaevskiy · 6 years ago
  2. 12abf67 Reland "Tune vp9 screenshare bitrate and framerate of spatial layers" by Ilya Nikolaevskiy · 6 years ago
  3. 304e9d2 Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current by Danil Chapovalov · 6 years ago
  4. 37d4f91 DCHECK feedback_rtt is positive by Evan Shrubsole · 6 years ago
  5. 184f6d5 Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video. by Rasmus Brandt · 6 years ago
  6. 159e53a Fix LibvpxVp8Encoder::FrameDropThreshold by Elad Alon · 6 years ago
  7. dac7aa0 Roll chromium_revision c71bb6f5f1..e328c33c20 (638607:638961) by chromium-webrtc-autoroll · 6 years ago
  8. f0cbcd3 Use stdlib TaskQueue implementation in webrtc fuzzers by Danil Chapovalov · 6 years ago
  9. 3caf50d Make ChangeBitrateVP9 unittest a bit more lenient. by Yves Gerey · 6 years ago
  10. 7f1c589 Adding new top-level directory crypto/ by Benjamin Wright · 6 years ago
  11. 1109b59 Revert "Tune vp9 screenshare bitrate and framerate of spatial layers" by Jeroen de Borst · 6 years ago
  12. 2c7b982 Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode" by Jeroen de Borst · 6 years ago
  13. 7e70291 Fix unscoped variable in test/scenario/BUILD.gn. by Mirko Bonadei · 6 years ago
  14. 74350db Roll chromium_revision 4205483be6..c71bb6f5f1 (638505:638607) by chromium-webrtc-autoroll · 6 years ago
  15. aaf3cb3 Tune vp9 screenshare bitrate and framerate of spatial layers by Ilya Nikolaevskiy · 6 years ago
  16. 39d3a7d Delete CodecSpecificInfo argument from VideoDecoder::Decode by Niels Möller · 6 years ago
  17. 1c90cab Fix UpdateRect handling for native buffers in VideoStreamEncoder by Ilya Nikolaevskiy · 6 years ago
  18. 6f0aafa Add PrintResults to VideoCodecTest. by Rasmus Brandt · 6 years ago
  19. d5af402 Add overhead observers to MediaTransportInterface by Niels Möller · 6 years ago
  20. 06b77f9 Use min allocatable bitrate as lower bound for target bitrate. by Sebastian Jansson · 6 years ago
  21. ffe9376 Bump iOS min supported version to 10.0 by Kári Tristan Helgason · 6 years ago
  22. b859b32 Update more VideoEncoder implementations to drop CodecSpecificInfo input by Niels Möller · 6 years ago
  23. 6318f13 Stop using rtc::TaskQueue::Current in RtcpTransceiver by Danil Chapovalov · 6 years ago
  24. dc62ae4 Cleanup of constraints configuration in GoogCcNetworkController. by Sebastian Jansson · 6 years ago
  25. 78b7d49 Roll chromium_revision 1af146a0f6..4205483be6 (638325:638505) by chromium-webrtc-autoroll · 6 years ago
  26. 8a1e35c Finally delete deprecated mac capturer. by Kári Tristan Helgason · 6 years ago
  27. 87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
  28. 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
  29. de3360e Create Vp8FrameBufferController by Elad Alon · 6 years ago
  30. 610c763 Add target bitrate headroom signal to VideoStreamEncoder. by Erik Språng · 6 years ago
  31. e49d64e Roll chromium_revision 3eb6e6ce76..1af146a0f6 (638159:638325) by chromium-webrtc-autoroll · 6 years ago
  32. 7276b97 Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC. by Benjamin Wright · 6 years ago
  33. 4423c36 Migrate RepeatingTask to take raw pointer to TaskQueueBase instead of TaskQueue by Danil Chapovalov · 6 years ago
  34. 11e55ee Renaming min_pacing_rate to min_total_allocated_bitrate. by Sebastian Jansson · 6 years ago
  35. 7b41225 Throttle frame-rate In VP8 encoder in steady state for screenshare by Ilya Nikolaevskiy · 6 years ago
  36. 2ecc8c8 Roll chromium_revision 99baeeafe2..3eb6e6ce76 (638035:638159) by chromium-webrtc-autoroll · 6 years ago
  37. 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
  38. ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
  39. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  40. 67f862e Guard against calls to OnEncodedFrame after Release. by Sami Kalliomäki · 6 years ago
  41. 6117068 Throttle frame-rate In VP9 encoder in steady state for screenshare by Ilya Nikolaevskiy · 6 years ago
  42. 0cb858c New VCMPacket constructor without WebRtcRTPHeader argument by Niels Möller · 6 years ago
  43. 7bc331f Android: Use android_deps directly by Peter Wen · 6 years ago
  44. c44f6cc Modernize RtpRtcp factory function: use unique_ptr as return type by Danil Chapovalov · 6 years ago
  45. ede7cb2 Rewrite video_loopback to use new mac capturer. by Kári Tristan Helgason · 6 years ago
  46. c8d2e73 Delete CodecSpecificInfo argument from VideoEncoder::Encode by Niels Möller · 6 years ago
  47. 1e42761 Removes verbose extension warning in Scenario tests. by Sebastian Jansson · 6 years ago
  48. 110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 6 years ago
  49. 745cfb9 use link_deps in ana_debug_dump_proto by Takuto Ikuta · 6 years ago
  50. d738071 Refactor FakeEncoder to avoid writing to a const EncodedImage by Niels Möller · 6 years ago
  51. 9df3353 Generic Frame Descriptor (GFD) VP8 templates. by philipel · 6 years ago
  52. 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
  53. 0da25a1 Update TransportSequenceNumberV2 extension to support fixed size by Johannes Kron · 6 years ago
  54. f441ea9 Minor cleanup of probe_controller.cc by Jonas Olsson · 6 years ago
  55. 200feba Make AEC3 the default desktop AEC option in WebRTC by Per Åhgren · 6 years ago
  56. 359c9b9 Roll chromium_revision 49f30ad2d0..99baeeafe2 (637641:638035) by chromium-webrtc-autoroll · 6 years ago
  57. be7af93 Add dsymutil as a mac cipd dependency. by Yves Gerey · 6 years ago
  58. b443dfe Use metrics::Samples in a couple pc/ tests by Steve Anton · 6 years ago
  59. e2a284d Adding metrics to measure usage of simulcast API. by Amit Hilbuch · 6 years ago
  60. 4eb5c14 Reland "Remove field trial include from decision logic." by Jakob Ivarsson · 6 years ago
  61. 07a4f2b Merge rtc_task_queue(_api|_impl)? build targets into one by Danil Chapovalov · 6 years ago
  62. 4450708 Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics. by Jakob Ivarsson · 6 years ago
  63. 1aa7581 Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask by Danil Chapovalov · 6 years ago
  64. c1e6e86 Add metrics::Samples to facilitate easier testing by Steve Anton · 6 years ago
  65. d36c086 Add support for simulcast streams in QualityAnalyzingVideoDecoder. by Artem Titov · 6 years ago
  66. 6ec2f54 Fix mis-spelled TODO items by Niels Möller · 6 years ago
  67. 7949f21 Revert "Removes lock from ChannelSend." by Sebastian Jansson · 6 years ago
  68. 9ef5e05 Fix target bitrate handling for a single layer VP9 screenshare by Ilya Nikolaevskiy · 6 years ago
  69. 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
  70. f23f161 Roll chromium_revision 5fe3fd14f6..49f30ad2d0 (637538:637641) by chromium-webrtc-autoroll · 6 years ago
  71. 423bae4 Revert "Remove field trial include from decision logic." by Jakob Ivarsson‎ · 6 years ago
  72. bd7ed4b Include sign for infinity in ToString for data units. by Sebastian Jansson · 6 years ago
  73. 44ce4b4 Adding a placeholder audio_buffer build target by Per Åhgren · 6 years ago
  74. 15845af Reland "Another mock for GetSctpTransport" (and add test) by Harald Alvestrand · 6 years ago
  75. 9b93447 Removes lock from ChannelSend. by Sebastian Jansson · 6 years ago
  76. c8eeb18 Fixes parsing of logs where receive time info is missing. by Sebastian Jansson · 6 years ago
  77. 07316a6 Roll chromium_revision 4229a4b64d..5fe3fd14f6 (637436:637538) by chromium-webrtc-autoroll · 6 years ago
  78. 1f8e445 Roll chromium_revision 5afa522447..4229a4b64d (637301:637436) by chromium-webrtc-autoroll · 6 years ago
  79. 572c60f Injecting Clock into video senders. by Sebastian Jansson · 6 years ago
  80. 8026d60 Injecting Clock in video receive. by Sebastian Jansson · 6 years ago
  81. ef50b25 Remove lock in WebRtcVideoEngine by Steve Anton · 6 years ago
  82. 4cde9ad Fix some typos found in ivf_file_writer.cc by Elad Alon · 6 years ago
  83. 4e5f5ed Allow Clock injection in Call. by Sebastian Jansson · 6 years ago
  84. 5fe9510 Move ownership of RTPSenderVideo one more level up, to RtpVideoSender by Niels Möller · 6 years ago
  85. ac6cf7f Roll chromium_revision e65d7afd91..5afa522447 (637200:637301) by chromium-webrtc-autoroll · 6 years ago
  86. da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
  87. d0f3d84 Wire UpdateRect signalling in test frame generators by Ilya Nikolaevskiy · 6 years ago
  88. acd8ae7 Reinstate old iceConnectionState "completed" behavior by Jonas Olsson · 6 years ago
  89. 0a16916 Use JavaAudioDeviceModule as default by Paulina Hensman · 6 years ago
  90. 13471a4 Switch back to native mutexes on macOS by Oskar Sundbom · 6 years ago
  91. b678940 Using send time instead of system clock in quality scaler. by Sebastian Jansson · 6 years ago
  92. e64a688 Replacing Clock in ScreenshareLayers. by Sebastian Jansson · 6 years ago
  93. c130d42 Add ability to unwind stack for the current thread by Karl Wiberg · 6 years ago
  94. 8b8d01a Add full stack test with weak 3g-like properties by Erik Språng · 6 years ago
  95. 727504c Revert "Another mock for GetSctpTransport" by Harald Alvestrand · 6 years ago
  96. 3b548dd Move rtc::NewClosure into own build target as ToQueuedTask by Danil Chapovalov · 6 years ago
  97. b2c4700 Another mock for GetSctpTransport by Harald Alvestrand · 6 years ago
  98. 87e05b5 NetEq fuzzer: lower the maximum fuzzer input size by Henrik Lundin · 6 years ago
  99. 7ceef35 Roll chromium_revision b3ef4b21cb..e65d7afd91 (637096:637200) by chromium-webrtc-autoroll · 6 years ago
  100. 4a42742 Make rtc_base/fake_mdns_responder.h self contained. by Mirko Bonadei · 6 years ago