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gerrit-public.fairphone.software / platform / external / webrtc / 22f9925b3eea255ea61edae6bd364249db2fc574
  1. 22f9925 webrtc: Remove semicolons. by Nico Weber · 5 years ago
  2. af623ae Delete unused file mock_video_codec_interface.h by Niels Möller · 5 years ago
  3. d36a815 Remove the deprecated CreateProbeClusters method by Piotr (Peter) Slatala · 5 years ago
  4. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 5 years ago
  5. 01fe309 Do not use RtcEventLogs in media transport when used only for data channel. by Piotr (Peter) Slatala · 5 years ago
  6. ce27875 [AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule by Alvaro Martinez · 5 years ago
  7. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 5 years ago
  8. d5e02f0 Delete redundant members from VCMPacket. by Niels Möller · 5 years ago
  9. 4d2367a Removes broken frame matching code in scenario quality stats. by Sebastian Jansson · 5 years ago
  10. b35bacc Fix NetEq minimum and maximum delay always reset on acm creation. by Ruslan Burakov · 5 years ago
  11. 8073db6 Roll chromium_revision 4b3282a5d6..554be8c5f4 (633587:633687) by chromium-webrtc-autoroll · 5 years ago
  12. 76d7ce2 Disabling flaky RecievesVp8SimulcastFrames test. by Sebastian Jansson · 5 years ago
  13. dd1cc98 Reland "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Möller · 5 years ago
  14. 109b5fb Revert "Extend TransportSequenceNumber RTP header extension" by Mirko Bonadei · 5 years ago
  15. 28c7362 Extend TransportSequenceNumber RTP header extension by Johannes Kron · 5 years ago
  16. 3f6bf3a Clarify that pacing rate is based on raw target rate by Evan Shrubsole · 5 years ago
  17. 5fbebd5 Adds support for VP8 simulcast to scenario tests. by Sebastian Jansson · 5 years ago
  18. ccb9b75 Create version 01 of Generic Frame Descriptor - with discardability flag by Elad Alon · 5 years ago
  19. 0b2150c Add a task queue into pc e2e fixture implementation by Artem Titov · 5 years ago
  20. e82643f Fix FFT output size to avoid incorrect band energy computation by Alessio Bazzica · 5 years ago
  21. cc26fef Use a CopyOnWriteBuffer to back EncodedImage data by Niels Möller · 5 years ago
  22. 0d4869c Roll chromium_revision d723882358..4b3282a5d6 (633435:633587) by chromium-webrtc-autoroll · 5 years ago
  23. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 5 years ago
  24. ba63caf Roll chromium_revision 086bdb74b2..d723882358 (633288:633435) by chromium-webrtc-autoroll · 5 years ago
  25. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 5 years ago
  26. 4e7058e desktopCaptuer: exempt to overlapping between hidden taskbar and maximized target by braveyao · 5 years ago
  27. 0e44907 Roll chromium_revision 55c117dd14..086bdb74b2 (633171:633288) by chromium-webrtc-autoroll · 5 years ago
  28. 7abfd56 Improve CPU utilization when encoding VP8 with two temporal layers by Elad Alon · 5 years ago
  29. 599d592 Extend RemoteEstimatorProxy to support feedback on sender request. by Johannes Kron · 5 years ago
  30. a89800c Parse params of 3rd spatial layer from command line. by Sergey Silkin · 5 years ago
  31. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 5 years ago
  32. 1925b5a Delete deprecated version of AudioCodingModule::IncomingPacket by Niels Möller · 5 years ago
  33. 1431572 Roll chromium_revision 0f484ff968..55c117dd14 (633071:633171) by chromium-webrtc-autoroll · 5 years ago
  34. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 5 years ago
  35. 7131880 Don't block the signaling thread during the call. by Mirko Bonadei · 5 years ago
  36. 0e1a1f9 Add verbose logging to encoder bitrate adjuster by Erik Språng · 5 years ago
  37. 4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 5 years ago
  38. 06c5145 Adds support for VP9 scalability layers to scenario tests. by Sebastian Jansson · 5 years ago
  39. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 5 years ago
  40. f2727fb Adds slides support to scenario tests. by Sebastian Jansson · 5 years ago
  41. e9652ca Android: Add video processing interface by Magnus Jedvert · 5 years ago
  42. 4a2d57a Don't include video_bitrate_allocation.h from encoded_image.h by Niels Möller · 5 years ago
  43. 71aee3a Reland "Propagate VideoFrame::UpdateRect to encoder" by Ilya Nikolaevskiy · 5 years ago
  44. f873cd9 Roll chromium_revision 26c36e3408..0f484ff968 (632825:633071) by chromium-webrtc-autoroll · 5 years ago
  45. bf47495 Update remaining audio test code to not use WebRtcRTPHeader. by Niels Möller · 5 years ago
  46. a0b1fb9 Pass H264 profile/level settings to codec. by Sergey Silkin · 5 years ago
  47. 3073c72 Fix AndroidVideoDecoderTest for new Robolectric version. by Sami Kalliomäki · 5 years ago
  48. e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 5 years ago
  49. d2f0436 Make sdk/android:{audio,video}_api_java publicly visible. by Mirko Bonadei · 5 years ago
  50. 0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 5 years ago
  51. 6e1402b Skip SSIM calculation in real time mode. by Sergey Silkin · 5 years ago
  52. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 5 years ago
  53. 389b167 Delete test/constants.h by Elad Alon · 5 years ago
  54. 8d2e228 Add thread safety annotations for PeerConnection::*_state_ by Karl Wiberg · 5 years ago
  55. e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 5 years ago
  56. eaf6a8c Adding src/third_party/androidx to the DEPS file. by Mirko Bonadei · 5 years ago
  57. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 5 years ago
  58. 86f0974 Roll chromium_revision 7df1a5ba86..26c36e3408 (632711:632825) by chromium-webrtc-autoroll · 5 years ago
  59. c664314 Clean up implementation in stream_params by Steve Anton · 5 years ago
  60. ca890ee Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Mirko Bonadei · 5 years ago
  61. ca3c801 Minor eventlogvisualizer tweaks. by Konrad Hofbauer · 5 years ago
  62. 429b67d Revert "Propagate VideoFrame::UpdateRect to encoder" by Mirko Bonadei · 5 years ago
  63. 675b47d Roll chromium_revision bf2d75ba40..7df1a5ba86 (632595:632711) by chromium-webrtc-autoroll · 5 years ago
  64. 9775a58 Plot bitrate allocation per layer based on RTCP XR target bitrate. by Bjorn Terelius · 5 years ago
  65. b03ab71 Add thread safety annotation for PeerConnection::event_log_ by Karl Wiberg · 5 years ago
  66. 744310f Add thread safety annotation for PeerConnection::observer_ and factory_ by Karl Wiberg · 5 years ago
  67. 7c974e6 Plot RTCP types for incoming and outgoing RTCP packets. by Bjorn Terelius · 5 years ago
  68. c39f462 Move RtcEventProbeClusterCreated to the network controller. by Piotr (Peter) Slatala · 5 years ago
  69. 6255af9 Fix RateCounter to don't fail if there are too small amount of events by Artem Titov · 5 years ago
  70. efa72a1 Propagate VideoFrame::UpdateRect to encoder by Ilya Nikolaevskiy · 5 years ago
  71. 3a656d1 Tune bitrates and minQP thresholds for high-fps screenshare. by Ilya Nikolaevskiy · 5 years ago
  72. c8221fc Roll chromium_revision d1f68eb66e..bf2d75ba40 (632477:632595) by chromium-webrtc-autoroll · 5 years ago
  73. 075f687 Add struct for feedback request to RTPHeaderExtension by Johannes Kron · 5 years ago
  74. 05d43c6 Fix getStats() freeze bug affecting Chromium but not WebRTC standalone. by Henrik Boström · 5 years ago
  75. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 5 years ago
  76. 106d92d Add thread safety annotation for PeerConnection::SignalDataChannelCreated_ by Karl Wiberg · 5 years ago
  77. 13bc871 PostMessageWithFunctor() added. by Henrik Boström · 5 years ago
  78. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 5 years ago
  79. 7e0e44f Move video-related MediaTransport interfaces to their own file and target by Niels Möller · 5 years ago
  80. 054db54 Remove an absl::WrapUnique usage without absl/memory/memory.h include by tzik · 5 years ago
  81. 22997d6 Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477) by chromium-webrtc-autoroll · 5 years ago
  82. 1c9c9fc Replace replace_substrs with Abseil by Steve Anton · 5 years ago
  83. bf9e01a Add support of fast media sending in peer connection e2e test by Artem Titov · 5 years ago
  84. ceba6ae Return a copy, becase GetPercentile in SamplesStatsCounter isn't const by Artem Titov · 5 years ago
  85. cf8405e Add generic packet rates to event_log_visualizer. by Piotr (Peter) Slatala · 5 years ago
  86. 15653f9 Roll chromium_revision 78de17c053..2ad52fb2a4 (632252:632357) by chromium-webrtc-autoroll · 5 years ago
  87. aa58415 Reland "Enabling Simulcast use via AddTransceiver." by Amit Hilbuch · 5 years ago
  88. aec9794 Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format. by Piotr (Peter) Slatala · 5 years ago
  89. 9e2692c Roll chromium_revision 9a34b2cc2d..78de17c053 (632146:632252) by chromium-webrtc-autoroll · 5 years ago
  90. d036c65 Clarify and unify outgoing and incoming packet loss rate plots. by Konrad Hofbauer · 5 years ago
  91. 663844d Update test code to use EncodedImage::Allocate by Niels Möller · 5 years ago
  92. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 5 years ago
  93. 92e7c69 Revert "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Moller · 5 years ago
  94. 8e847ee Make recv_deltas optional in TransportFeedback packets by Johannes Kron · 5 years ago
  95. 69fb6c8 Allow DtlsTransport::Information() to be called off-thread by Harald Alvestrand · 5 years ago
  96. 068fc35 Break out parameters from EventLogAnalyzer to AnalyzerConfig struct. by Bjorn Terelius · 5 years ago
  97. f0c366b Cleanup of scenario test video stream setup. by Sebastian Jansson · 5 years ago
  98. d00045e Changing command line flag for scenario logs root directory. by Sebastian Jansson · 5 years ago
  99. dac03d9 Move MediaConstraintsInterface to sdk/, and make it a concrete class by Niels Möller · 5 years ago
  100. 1d7bf89 Add LS_VERBOSE logging for target bitrate in GoogCC by Evan Shrubsole · 5 years ago