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gerrit-public.fairphone.software / platform / external / webrtc / b7cb7b5e944630c0b4c1412a41622c6bea1efecc
  1. b7cb7b5 Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder by Erik Språng · 6 years ago
  2. 695af94 Add reentrancy comment for critical section. by Ruslan Burakov · 6 years ago
  3. fee13e8 Log pacer values to verbose log by Evan Shrubsole · 6 years ago
  4. 12ae4f4 Introduce possibility to poll stats and notify analyzers. by Mirko Bonadei · 6 years ago
  5. 2684ab3 Test default TaskQueue implementation via TaskQueueBase interface by Danil Chapovalov · 6 years ago
  6. 22dab11 Remove Legacy ADM from AppRTC mobile by Paulina Hensman · 6 years ago
  7. 0bf4c29 Add support of auto IP generation in network emulation manager. by Artem Titov · 6 years ago
  8. 9595d1b Roll chromium_revision 15651144f3..ec3bf6e607 (635345:635450) by chromium-webrtc-autoroll · 6 years ago
  9. e2da931 Remove a leftover audio codec poison immutinty declaration by Karl Wiberg · 6 years ago
  10. f2889bb Add option to inject YuvConverter to SurfaceTextureHelper. by Åsa Persson · 6 years ago
  11. b4f0393 Roll chromium_revision a55c7bb989..15651144f3 (635189:635345) by chromium-webrtc-autoroll · 6 years ago
  12. bd0deca Ban absl::StrSplit and absl::StrJoin by Karl Wiberg · 6 years ago
  13. 7572bb4 Fix -Wextra-semi warnings in webrtc fuzzers. by Nico Weber · 6 years ago
  14. c35a72c Roll chromium_revision 81fda909f3..a55c7bb989 (635067:635189) by chromium-webrtc-autoroll · 6 years ago
  15. b000b71 Wiring up RIDs from the video engine to the RTP Sender. by Amit Hilbuch · 6 years ago
  16. 98335f8 Fixing webrtc::IceTransportState. by Seth Hampson · 6 years ago
  17. 5cbc528 Revert "Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder" by Sami Kalliomäki · 6 years ago
  18. 7d6a4c0 Connect LossNotificationController to RtpRtcp by Elad Alon · 6 years ago
  19. a497d12 Avoids PostTask to repost a repeated task. by Sebastian Jansson · 6 years ago
  20. ce7a4fb Adding possibility to save an RTCEventLog of the call. by Mirko Bonadei · 6 years ago
  21. 99f5d5f Roll chromium_revision 95a23eca14..81fda909f3 (634895:635067) by chromium-webrtc-autoroll · 6 years ago
  22. d37307c Reland "Adds resource path support for video files in scenario tests." by Sebastian Jansson · 6 years ago
  23. 715c476 Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder by Erik Språng · 6 years ago
  24. 2b08e31 Adds CoDel implementation to network simulation. by Sebastian Jansson · 6 years ago
  25. 418dd0b Stop using special RTT value for DelayBasedBwe. by Sebastian Jansson · 6 years ago
  26. 76a74e6 Delay bug during audio receiver stream recreation. by Ruslan Burakov · 6 years ago
  27. c4dd730 Fix -Wextra-semi warnings. by Mirko Bonadei · 6 years ago
  28. 3812fa9 Delete VideoCodecTestParameterized. by Rasmus Brandt · 6 years ago
  29. 19d0104 Make RtpRtcp::Configuration::field_trials ptr const by Per Kjellander · 6 years ago
  30. a9cfa47 Revert "Delete rtc_task_queue_impl build target" by Mirko Bonadei · 6 years ago
  31. 74f0a51 Move kFeedbackMessageType from Remb to Psfb by Elad Alon · 6 years ago
  32. 56973e6 Delete rtc_task_queue_impl build target by Danil Chapovalov · 6 years ago
  33. 8721bb3 Roll chromium_revision e7ecd1bfc2..95a23eca14 (634731:634895) by chromium-webrtc-autoroll · 6 years ago
  34. e1e789b Removing non-const RtpSenderInterface::GetParameters(). by Amit Hilbuch · 6 years ago
  35. f58e43e Add an OpenChannel method to MediaTransportInterface and call it whenever PeerConnection opens a new data channel. by Bjorn Mellem · 6 years ago
  36. 8f096d0 Map clat devices to cellular on Android by Jeroen de Borst · 6 years ago
  37. e19a6da Roll chromium_revision a77f654a3c..e7ecd1bfc2 (634608:634731) by chromium-webrtc-autoroll · 6 years ago
  38. 487c09b Adds FakeNetworkPipeTest to rtc_unittests. by Sebastian Jansson · 6 years ago
  39. 29f9cd9 Synchronize replaceRegion calls. by Anders Carlsson · 6 years ago
  40. 7ef34f8 Replace field trials with WebRtcKeyValueConfig in PacedSender by Per Kjellander · 6 years ago
  41. ce8e867 Add support for TransportSequenceNumberV2 in SDP negotiation by Johannes Kron · 6 years ago
  42. 14f96d1 Roll chromium_revision f39a1b8992..a77f654a3c (634190:634608) by chromium-webrtc-autoroll · 6 years ago
  43. 8aa00f0 Add missing absl/memory/memory.h to rtc_event_generic_ack_received.cc by tzik · 6 years ago
  44. b4643ad Rename "OnReceivedFrame" to "OnAssembledFrame" by Elad Alon · 6 years ago
  45. d7329ca Remove VideoSender and fold code into VideoStreamEncoder by Erik Språng · 6 years ago
  46. 10874b2 Create LossNotificationController by Elad Alon · 6 years ago
  47. b75d9e9 Allow IceConnectionState to become failed without ever connecting. by Jonas Olsson · 6 years ago
  48. d209cd1 Lower SSIM thresholds. by Sergey Silkin · 6 years ago
  49. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  50. caa499b PFFFT C++ wrapper for APM by Alessio Bazzica · 6 years ago
  51. 45af00f Revert "Adds resource path support for video files in scenario tests." by Sergey Silkin · 6 years ago
  52. 4ae6347 Use `final` so that the compiler will be able to inline calls by Karl Wiberg · 6 years ago
  53. 5966c50 Add thread safety annotations for PeerConnection::configuration_ by Karl Wiberg · 6 years ago
  54. 8306a73 Adds resource path support for video files in scenario tests. by Sebastian Jansson · 6 years ago
  55. 96fccfe Make sure RTC_SUPPORTS_METAL is set in AppRTCMobile. by Anders Carlsson · 6 years ago
  56. 735f823 CreateAudioProcessor: do not propagate an unset echo control factory to the AudioProcessing instance by Jesús de Vicente Peña · 6 years ago
  57. bed8604 Adding entry point for the v2 stats API. by Peter Hanspers · 6 years ago
  58. 2645193 DtlsTransport::ice_transport is const and can be called off thread by Harald Alvestrand · 6 years ago
  59. ee95f3e Roll chromium_revision 94ca2b10d8..f39a1b8992 (634089:634190) by chromium-webrtc-autoroll · 6 years ago
  60. 54047be Reland "Extend TransportSequenceNumber RTP header extension" by Johannes Kron · 6 years ago
  61. 1eb3d7e Refactor DelayManager into separate Histogram class and make it injectable for testing purposes. by Jakob Ivarsson · 6 years ago
  62. fa52efa Migrate stdlib task queue to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  63. e11b7d2 Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule by Per Kjellander · 6 years ago
  64. aa1a43e AEC3: Use minimum ERLE during onsets by Gustaf Ullberg · 6 years ago
  65. d6c6f16 Update RTP packet and header fuzzers to support additional extensions by Johannes Kron · 6 years ago
  66. 3256225 "Remove" loophole in rtc::Thread::ScopedDisallowBlockingCalls by Karl Wiberg · 6 years ago
  67. 826f2e7 Migrate win task queue to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  68. bb05369 Delete unused class FakeCandidatePair. by Niels Möller · 6 years ago
  69. 00c57e3 Delete unused class RtpTransportInternalAdapter by Niels Möller · 6 years ago
  70. 17c147c Feed PacedSender with RTP packet size by Per Kjellander · 6 years ago
  71. 252725d Rename RtpPacketHistory::PacketState::payload_size -> packet_size by Per Kjellander · 6 years ago
  72. 1b801e0 Roll chromium_revision 55c441e653..94ca2b10d8 (633987:634089) by chromium-webrtc-autoroll · 6 years ago
  73. feef8f5 Roll chromium_revision 919d2e8241..55c441e653 (633811:633987) by chromium-webrtc-autoroll · 6 years ago
  74. 32232e9 Add spatial layers support to video analyze pipeline. by Artem Titov · 6 years ago
  75. 8e68920 Roll chromium_revision 554be8c5f4..919d2e8241 (633687:633811) by chromium-webrtc-autoroll · 6 years ago
  76. 47cf5ea Migrate gcd task queue implementation to TaskQueueBase interface by Danil Chapovalov · 6 years ago
  77. f5d8808 Remove Analyzers struct. by Mirko Bonadei · 6 years ago
  78. 22f9925 webrtc: Remove semicolons. by Nico Weber · 6 years ago
  79. af623ae Delete unused file mock_video_codec_interface.h by Niels Möller · 6 years ago
  80. d36a815 Remove the deprecated CreateProbeClusters method by Piotr (Peter) Slatala · 6 years ago
  81. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  82. 01fe309 Do not use RtcEventLogs in media transport when used only for data channel. by Piotr (Peter) Slatala · 6 years ago
  83. ce27875 [AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule by Alvaro Martinez · 6 years ago
  84. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  85. d5e02f0 Delete redundant members from VCMPacket. by Niels Möller · 6 years ago
  86. 4d2367a Removes broken frame matching code in scenario quality stats. by Sebastian Jansson · 6 years ago
  87. b35bacc Fix NetEq minimum and maximum delay always reset on acm creation. by Ruslan Burakov · 6 years ago
  88. 8073db6 Roll chromium_revision 4b3282a5d6..554be8c5f4 (633587:633687) by chromium-webrtc-autoroll · 6 years ago
  89. 76d7ce2 Disabling flaky RecievesVp8SimulcastFrames test. by Sebastian Jansson · 6 years ago
  90. dd1cc98 Reland "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Möller · 6 years ago
  91. 109b5fb Revert "Extend TransportSequenceNumber RTP header extension" by Mirko Bonadei · 6 years ago
  92. 28c7362 Extend TransportSequenceNumber RTP header extension by Johannes Kron · 6 years ago
  93. 3f6bf3a Clarify that pacing rate is based on raw target rate by Evan Shrubsole · 6 years ago
  94. 5fbebd5 Adds support for VP8 simulcast to scenario tests. by Sebastian Jansson · 6 years ago
  95. ccb9b75 Create version 01 of Generic Frame Descriptor - with discardability flag by Elad Alon · 6 years ago
  96. 0b2150c Add a task queue into pc e2e fixture implementation by Artem Titov · 6 years ago
  97. e82643f Fix FFT output size to avoid incorrect band energy computation by Alessio Bazzica · 6 years ago
  98. cc26fef Use a CopyOnWriteBuffer to back EncodedImage data by Niels Möller · 6 years ago
  99. 0d4869c Roll chromium_revision d723882358..4b3282a5d6 (633435:633587) by chromium-webrtc-autoroll · 6 years ago
  100. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 6 years ago