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gerrit-public.fairphone.software / platform / external / webrtc / f58e43e2a69764702016d3fa4358a53257a5662c
  1. f58e43e Add an OpenChannel method to MediaTransportInterface and call it whenever PeerConnection opens a new data channel. by Bjorn Mellem · 6 years ago
  2. 8f096d0 Map clat devices to cellular on Android by Jeroen de Borst · 6 years ago
  3. e19a6da Roll chromium_revision a77f654a3c..e7ecd1bfc2 (634608:634731) by chromium-webrtc-autoroll · 6 years ago
  4. 487c09b Adds FakeNetworkPipeTest to rtc_unittests. by Sebastian Jansson · 6 years ago
  5. 29f9cd9 Synchronize replaceRegion calls. by Anders Carlsson · 6 years ago
  6. 7ef34f8 Replace field trials with WebRtcKeyValueConfig in PacedSender by Per Kjellander · 6 years ago
  7. ce8e867 Add support for TransportSequenceNumberV2 in SDP negotiation by Johannes Kron · 6 years ago
  8. 14f96d1 Roll chromium_revision f39a1b8992..a77f654a3c (634190:634608) by chromium-webrtc-autoroll · 6 years ago
  9. 8aa00f0 Add missing absl/memory/memory.h to rtc_event_generic_ack_received.cc by tzik · 6 years ago
  10. b4643ad Rename "OnReceivedFrame" to "OnAssembledFrame" by Elad Alon · 6 years ago
  11. d7329ca Remove VideoSender and fold code into VideoStreamEncoder by Erik Språng · 6 years ago
  12. 10874b2 Create LossNotificationController by Elad Alon · 6 years ago
  13. b75d9e9 Allow IceConnectionState to become failed without ever connecting. by Jonas Olsson · 6 years ago
  14. d209cd1 Lower SSIM thresholds. by Sergey Silkin · 6 years ago
  15. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  16. caa499b PFFFT C++ wrapper for APM by Alessio Bazzica · 6 years ago
  17. 45af00f Revert "Adds resource path support for video files in scenario tests." by Sergey Silkin · 6 years ago
  18. 4ae6347 Use `final` so that the compiler will be able to inline calls by Karl Wiberg · 6 years ago
  19. 5966c50 Add thread safety annotations for PeerConnection::configuration_ by Karl Wiberg · 6 years ago
  20. 8306a73 Adds resource path support for video files in scenario tests. by Sebastian Jansson · 6 years ago
  21. 96fccfe Make sure RTC_SUPPORTS_METAL is set in AppRTCMobile. by Anders Carlsson · 6 years ago
  22. 735f823 CreateAudioProcessor: do not propagate an unset echo control factory to the AudioProcessing instance by Jesús de Vicente Peña · 6 years ago
  23. bed8604 Adding entry point for the v2 stats API. by Peter Hanspers · 6 years ago
  24. 2645193 DtlsTransport::ice_transport is const and can be called off thread by Harald Alvestrand · 6 years ago
  25. ee95f3e Roll chromium_revision 94ca2b10d8..f39a1b8992 (634089:634190) by chromium-webrtc-autoroll · 6 years ago
  26. 54047be Reland "Extend TransportSequenceNumber RTP header extension" by Johannes Kron · 6 years ago
  27. 1eb3d7e Refactor DelayManager into separate Histogram class and make it injectable for testing purposes. by Jakob Ivarsson · 6 years ago
  28. fa52efa Migrate stdlib task queue to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  29. e11b7d2 Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule by Per Kjellander · 6 years ago
  30. aa1a43e AEC3: Use minimum ERLE during onsets by Gustaf Ullberg · 6 years ago
  31. d6c6f16 Update RTP packet and header fuzzers to support additional extensions by Johannes Kron · 6 years ago
  32. 3256225 "Remove" loophole in rtc::Thread::ScopedDisallowBlockingCalls by Karl Wiberg · 6 years ago
  33. 826f2e7 Migrate win task queue to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  34. bb05369 Delete unused class FakeCandidatePair. by Niels Möller · 6 years ago
  35. 00c57e3 Delete unused class RtpTransportInternalAdapter by Niels Möller · 6 years ago
  36. 17c147c Feed PacedSender with RTP packet size by Per Kjellander · 6 years ago
  37. 252725d Rename RtpPacketHistory::PacketState::payload_size -> packet_size by Per Kjellander · 6 years ago
  38. 1b801e0 Roll chromium_revision 55c441e653..94ca2b10d8 (633987:634089) by chromium-webrtc-autoroll · 6 years ago
  39. feef8f5 Roll chromium_revision 919d2e8241..55c441e653 (633811:633987) by chromium-webrtc-autoroll · 6 years ago
  40. 32232e9 Add spatial layers support to video analyze pipeline. by Artem Titov · 6 years ago
  41. 8e68920 Roll chromium_revision 554be8c5f4..919d2e8241 (633687:633811) by chromium-webrtc-autoroll · 6 years ago
  42. 47cf5ea Migrate gcd task queue implementation to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  43. f5d8808 Remove Analyzers struct. by Mirko Bonadei · 6 years ago
  44. 22f9925 webrtc: Remove semicolons. by Nico Weber · 6 years ago
  45. af623ae Delete unused file mock_video_codec_interface.h by Niels Möller · 6 years ago
  46. d36a815 Remove the deprecated CreateProbeClusters method by Piotr (Peter) Slatala · 6 years ago
  47. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  48. 01fe309 Do not use RtcEventLogs in media transport when used only for data channel. by Piotr (Peter) Slatala · 6 years ago
  49. ce27875 [AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule by Alvaro Martinez · 6 years ago
  50. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  51. d5e02f0 Delete redundant members from VCMPacket. by Niels Möller · 6 years ago
  52. 4d2367a Removes broken frame matching code in scenario quality stats. by Sebastian Jansson · 6 years ago
  53. b35bacc Fix NetEq minimum and maximum delay always reset on acm creation. by Ruslan Burakov · 6 years ago
  54. 8073db6 Roll chromium_revision 4b3282a5d6..554be8c5f4 (633587:633687) by chromium-webrtc-autoroll · 6 years ago
  55. 76d7ce2 Disabling flaky RecievesVp8SimulcastFrames test. by Sebastian Jansson · 6 years ago
  56. dd1cc98 Reland "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Möller · 6 years ago
  57. 109b5fb Revert "Extend TransportSequenceNumber RTP header extension" by Mirko Bonadei · 6 years ago
  58. 28c7362 Extend TransportSequenceNumber RTP header extension by Johannes Kron · 6 years ago
  59. 3f6bf3a Clarify that pacing rate is based on raw target rate by Evan Shrubsole · 6 years ago
  60. 5fbebd5 Adds support for VP8 simulcast to scenario tests. by Sebastian Jansson · 6 years ago
  61. ccb9b75 Create version 01 of Generic Frame Descriptor - with discardability flag by Elad Alon · 6 years ago
  62. 0b2150c Add a task queue into pc e2e fixture implementation by Artem Titov · 6 years ago
  63. e82643f Fix FFT output size to avoid incorrect band energy computation by Alessio Bazzica · 6 years ago
  64. cc26fef Use a CopyOnWriteBuffer to back EncodedImage data by Niels Möller · 6 years ago
  65. 0d4869c Roll chromium_revision d723882358..4b3282a5d6 (633435:633587) by chromium-webrtc-autoroll · 6 years ago
  66. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 6 years ago
  67. ba63caf Roll chromium_revision 086bdb74b2..d723882358 (633288:633435) by chromium-webrtc-autoroll · 6 years ago
  68. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
  69. 4e7058e desktopCaptuer: exempt to overlapping between hidden taskbar and maximized target by braveyao · 6 years ago
  70. 0e44907 Roll chromium_revision 55c117dd14..086bdb74b2 (633171:633288) by chromium-webrtc-autoroll · 6 years ago
  71. 7abfd56 Improve CPU utilization when encoding VP8 with two temporal layers by Elad Alon · 6 years ago
  72. 599d592 Extend RemoteEstimatorProxy to support feedback on sender request. by Johannes Kron · 6 years ago
  73. a89800c Parse params of 3rd spatial layer from command line. by Sergey Silkin · 6 years ago
  74. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 6 years ago
  75. 1925b5a Delete deprecated version of AudioCodingModule::IncomingPacket by Niels Möller · 6 years ago
  76. 1431572 Roll chromium_revision 0f484ff968..55c117dd14 (633071:633171) by chromium-webrtc-autoroll · 6 years ago
  77. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 6 years ago
  78. 7131880 Don't block the signaling thread during the call. by Mirko Bonadei · 6 years ago
  79. 0e1a1f9 Add verbose logging to encoder bitrate adjuster by Erik Språng · 6 years ago
  80. 4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 6 years ago
  81. 06c5145 Adds support for VP9 scalability layers to scenario tests. by Sebastian Jansson · 6 years ago
  82. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 6 years ago
  83. f2727fb Adds slides support to scenario tests. by Sebastian Jansson · 6 years ago
  84. e9652ca Android: Add video processing interface by Magnus Jedvert · 6 years ago
  85. 4a2d57a Don't include video_bitrate_allocation.h from encoded_image.h by Niels Möller · 6 years ago
  86. 71aee3a Reland "Propagate VideoFrame::UpdateRect to encoder" by Ilya Nikolaevskiy · 6 years ago
  87. f873cd9 Roll chromium_revision 26c36e3408..0f484ff968 (632825:633071) by chromium-webrtc-autoroll · 6 years ago
  88. bf47495 Update remaining audio test code to not use WebRtcRTPHeader. by Niels Möller · 6 years ago
  89. a0b1fb9 Pass H264 profile/level settings to codec. by Sergey Silkin · 6 years ago
  90. 3073c72 Fix AndroidVideoDecoderTest for new Robolectric version. by Sami Kalliomäki · 6 years ago
  91. e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 6 years ago
  92. d2f0436 Make sdk/android:{audio,video}_api_java publicly visible. by Mirko Bonadei · 6 years ago
  93. 0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 6 years ago
  94. 6e1402b Skip SSIM calculation in real time mode. by Sergey Silkin · 6 years ago
  95. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 6 years ago
  96. 389b167 Delete test/constants.h by Elad Alon · 6 years ago
  97. 8d2e228 Add thread safety annotations for PeerConnection::*_state_ by Karl Wiberg · 6 years ago
  98. e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 6 years ago
  99. eaf6a8c Adding src/third_party/androidx to the DEPS file. by Mirko Bonadei · 6 years ago
  100. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 6 years ago