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  1. 7abfd56 Improve CPU utilization when encoding VP8 with two temporal layers by Elad Alon · 57 minutes ago master
  2. 599d592 Extend RemoteEstimatorProxy to support feedback on sender request. by Johannes Kron · 2 hours ago lkgr
  3. a89800c Parse params of 3rd spatial layer from command line. by Sergey Silkin · 3 hours ago
  4. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 7 hours ago
  5. 1925b5a1 Delete deprecated version of AudioCodingModule::IncomingPacket by Niels Möller · 7 hours ago
  6. 1431572 Roll chromium_revision 0f484ff968..55c117dd14 (633071:633171) by chromium-webrtc-autoroll · 15 hours ago
  7. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 17 hours ago
  8. 7131880 Don't block the signaling thread during the call. by Mirko Bonadei · 22 hours ago
  9. 0e1a1f9 Add verbose logging to encoder bitrate adjuster by Erik Språng · 22 hours ago
  10. 4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 22 hours ago
  11. 06c5145 Adds support for VP9 scalability layers to scenario tests. by Sebastian Jansson · 22 hours ago
  12. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 23 hours ago
  13. f2727fb Adds slides support to scenario tests. by Sebastian Jansson · 24 hours ago
  14. e9652ca Android: Add video processing interface by Magnus Jedvert · 24 hours ago
  15. 4a2d57a Don't include video_bitrate_allocation.h from encoded_image.h by Niels Möller · 26 hours ago
  16. 71aee3a Reland "Propagate VideoFrame::UpdateRect to encoder" by Ilya Nikolaevskiy · 26 hours ago
  17. f873cd9 Roll chromium_revision 26c36e3408..0f484ff968 (632825:633071) by chromium-webrtc-autoroll · 26 hours ago
  18. bf47495 Update remaining audio test code to not use WebRtcRTPHeader. by Niels Möller · 27 hours ago
  19. a0b1fb9 Pass H264 profile/level settings to codec. by Sergey Silkin · 27 hours ago
  20. 3073c72 Fix AndroidVideoDecoderTest for new Robolectric version. by Sami Kalliomäki · 27 hours ago
  21. e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 30 hours ago
  22. d2f0436 Make sdk/android:{audio,video}_api_java publicly visible. by Mirko Bonadei · 31 hours ago
  23. 0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 31 hours ago
  24. 6e1402b Skip SSIM calculation in real time mode. by Sergey Silkin · 31 hours ago
  25. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 32 hours ago
  26. 389b167 Delete test/constants.h by Elad Alon · 2 days ago
  27. 8d2e228 Add thread safety annotations for PeerConnection::*_state_ by Karl Wiberg · 2 days ago
  28. e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 3 days ago
  29. eaf6a8c Adding src/third_party/androidx to the DEPS file. by Mirko Bonadei · 3 days ago
  30. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 4 days ago
  31. 86f0974 Roll chromium_revision 7df1a5ba86..26c36e3408 (632711:632825) by chromium-webrtc-autoroll · 4 days ago
  32. c664314 Clean up implementation in stream_params by Steve Anton · 4 days ago
  33. ca890ee Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Mirko Bonadei · 4 days ago
  34. ca3c801 Minor eventlogvisualizer tweaks. by Konrad Hofbauer · 4 days ago
  35. 429b67d Revert "Propagate VideoFrame::UpdateRect to encoder" by Mirko Bonadei · 4 days ago
  36. 675b47d Roll chromium_revision bf2d75ba40..7df1a5ba86 (632595:632711) by chromium-webrtc-autoroll · 4 days ago
  37. 9775a58 Plot bitrate allocation per layer based on RTCP XR target bitrate. by Bjorn Terelius · 4 days ago
  38. b03ab71 Add thread safety annotation for PeerConnection::event_log_ by Karl Wiberg · 4 days ago
  39. 744310f Add thread safety annotation for PeerConnection::observer_ and factory_ by Karl Wiberg · 4 days ago
  40. 7c974e6 Plot RTCP types for incoming and outgoing RTCP packets. by Bjorn Terelius · 4 days ago
  41. c39f462 Move RtcEventProbeClusterCreated to the network controller. by Piotr (Peter) Slatala · 4 days ago
  42. 6255af9 Fix RateCounter to don't fail if there are too small amount of events by Artem Titov · 4 days ago
  43. efa72a1 Propagate VideoFrame::UpdateRect to encoder by Ilya Nikolaevskiy · 4 days ago
  44. 3a656d1 Tune bitrates and minQP thresholds for high-fps screenshare. by Ilya Nikolaevskiy · 4 days ago
  45. c8221fc Roll chromium_revision d1f68eb66e..bf2d75ba40 (632477:632595) by chromium-webrtc-autoroll · 4 days ago
  46. 075f687 Add struct for feedback request to RTPHeaderExtension by Johannes Kron · 4 days ago
  47. 05d43c6 Fix getStats() freeze bug affecting Chromium but not WebRTC standalone. by Henrik Boström · 4 days ago
  48. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 4 days ago
  49. 106d92d Add thread safety annotation for PeerConnection::SignalDataChannelCreated_ by Karl Wiberg · 4 days ago
  50. 13bc871 PostMessageWithFunctor() added. by Henrik Boström · 4 days ago
  51. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 4 days ago
  52. 7e0e44f Move video-related MediaTransport interfaces to their own file and target by Niels Möller · 4 days ago
  53. 054db54 Remove an absl::WrapUnique usage without absl/memory/memory.h include by tzik · 4 days ago
  54. 22997d6 Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477) by chromium-webrtc-autoroll · 5 days ago
  55. 1c9c9fc Replace replace_substrs with Abseil by Steve Anton · 5 days ago
  56. bf9e01a Add support of fast media sending in peer connection e2e test by Artem Titov · 5 days ago
  57. ceba6ae Return a copy, becase GetPercentile in SamplesStatsCounter isn't const by Artem Titov · 5 days ago
  58. cf8405e Add generic packet rates to event_log_visualizer. by Piotr (Peter) Slatala · 5 days ago
  59. 15653f9 Roll chromium_revision 78de17c053..2ad52fb2a4 (632252:632357) by chromium-webrtc-autoroll · 5 days ago
  60. aa58415 Reland "Enabling Simulcast use via AddTransceiver." by Amit Hilbuch · 5 days ago
  61. aec9794 Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format. by Piotr (Peter) Slatala · 5 days ago
  62. 9e2692c Roll chromium_revision 9a34b2cc2d..78de17c053 (632146:632252) by chromium-webrtc-autoroll · 5 days ago
  63. d036c65 Clarify and unify outgoing and incoming packet loss rate plots. by Konrad Hofbauer · 5 days ago
  64. 663844d Update test code to use EncodedImage::Allocate by Niels Möller · 5 days ago
  65. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 5 days ago
  66. 92e7c69 Revert "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Moller · 5 days ago
  67. 8e847ee Make recv_deltas optional in TransportFeedback packets by Johannes Kron · 5 days ago
  68. 69fb6c8 Allow DtlsTransport::Information() to be called off-thread by Harald Alvestrand · 5 days ago
  69. 068fc35 Break out parameters from EventLogAnalyzer to AnalyzerConfig struct. by Bjorn Terelius · 5 days ago
  70. f0c366b Cleanup of scenario test video stream setup. by Sebastian Jansson · 5 days ago
  71. d00045e Changing command line flag for scenario logs root directory. by Sebastian Jansson · 5 days ago
  72. dac03d9 Move MediaConstraintsInterface to sdk/, and make it a concrete class by Niels Möller · 5 days ago
  73. 1d7bf89 Add LS_VERBOSE logging for target bitrate in GoogCC by Evan Shrubsole · 5 days ago
  74. 0179a3d Roll chromium_revision bfd7bcf815..9a34b2cc2d (632040:632146) by chromium-webrtc-autoroll · 5 days ago
  75. da825b1 Replace NOTREACHED with a break. by Piotr (Peter) Slatala · 5 days ago
  76. 1290fc7 Remove old accessor in GenericAckReceived by Piotr (Peter) Slatala · 5 days ago
  77. 788f577 Update the resolution check for VP8 simulcast. by Mirta Dvornicic · 5 days ago
  78. 6b88a8f Introduce default video quality analyzer by Artem Titov · 5 days ago
  79. b1ea48c Roll chromium_revision 103665932a..bfd7bcf815 (631883:632040) by chromium-webrtc-autoroll · 6 days ago
  80. 5ae259e Use a provider in rtc::Network to access the mDNS responder. by Qingsi Wang · 6 days ago
  81. 616b233 Add FullStackTest with simulated encoder overshooting by Erik Språng · 6 days ago
  82. 6c02541 Revert "Delete video source proxying in WebRtcVideoSendStream" by Christian Fremerey · 6 days ago
  83. 3588394 Roll chromium_revision d026ac796d..103665932a (631722:631883) by chromium-webrtc-autoroll · 6 days ago
  84. dfd5c4b Parse XR, FIR and PLI in rtc_event_log_parser.cc by Bjorn Terelius · 6 days ago
  85. 3c119fb Handle HKDF key derivation when building with OpenSSL. by Sergey Sablin · 6 days ago
  86. 5e2aad1 Support GenericPacketReceived/Sent/AckReceived event logs. by Piotr (Peter) Slatala · 6 days ago
  87. 975a899 Roll chromium_revision aa7b61fdc4..d026ac796d (631597:631722) by chromium-webrtc-autoroll · 6 days ago
  88. 4a68fb9 Separate base minimum delay and minimum delay. by Ruslan Burakov · 6 days ago
  89. 69bb3af Update EncodedFrameForMediaTransport to use Retain() rather than set_buffer + memcpy. by Niels Möller · 6 days ago
  90. 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 days ago
  91. cbf5949 Update MultiplexEncoderAdapter to use EncodedImage::Allocate by Niels Möller · 6 days ago
  92. 448c387 IceTransportWithTransportChannel: Initialize |thread_checker_| in declaration by Raphael Kubo da Costa · 6 days ago
  93. 2bd54a1 Ensure TestPeers are destroyed at the end of Run. by Mirko Bonadei · 6 days ago
  94. 6aca0b7 Add |update_rect| field and UpdateRect struct to VideoFrame. by Ilya Nikolaevskiy · 6 days ago
  95. 7f24fb9 Add settings to turn off VP8 base layer qp limit by Erik Språng · 6 days ago
  96. 98bcd32 Remove always_passing_unittest.cc. by Mirko Bonadei · 6 days ago
  97. b4f7ab1 Fix -Wunused-result warnings by Hans Wennborg · 6 days ago
  98. eedb0a1 Roll chromium_revision 23b4d2134b..aa7b61fdc4 (631425:631597) by chromium-webrtc-autoroll · 6 days ago
  99. a795c3b Roll chromium_revision d366835eb8..23b4d2134b (631269:631425) by chromium-webrtc-autoroll · 7 days ago
  100. dcbdd2c Add Foundation.framework to cocoa_threading target by Jiawei Ou · 7 days ago