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gerrit-public.fairphone.software / platform / external / webrtc / 82ce384801d2002b030b88dc1c30b8b2f5a29bc2
  1. 82ce384 Add improvement directions to PC and Call framework metrics by Artem Titov · 5 years ago
  2. 834a554 Include module_common_types.h only where needed by Niels Möller · 5 years ago
  3. bf5ee00 Disable prerender smoothing in MultiCodecReceiveTest. by Åsa Persson · 5 years ago
  4. a8e6f34 Delete the BasicPortAllocator constructor that enables gturn by Niels Möller · 5 years ago
  5. f2690a1 Delete unused method SendSideBandwidthEstimation::UpdateReceiverBlock by Niels Möller · 5 years ago
  6. bc646ee Roll chromium_revision 09b71d3027..1d4ed9e21d (698937:699120) by chromium-webrtc-autoroll · 5 years ago
  7. cd40de9 Delete the deprecated GetTransportParametersOffer(). by Bjorn A Mellem · 5 years ago
  8. 988e63e Proxy OnRtcpPacketReceived to the worker thread in channel tests. by Bjorn A Mellem · 5 years ago
  9. aab43db Roll chromium_revision 82de2e611e..09b71d3027 (698813:698937) by chromium-webrtc-autoroll · 5 years ago
  10. a99b89b AEC3: Echo remover handles multiple capture signals. by Gustaf Ullberg · 5 years ago
  11. 3433d56 Reduce resolution and bitrates of smoke test by Johannes Kron · 5 years ago
  12. f7457e5 Store PacketBuffer by value instead of as reference counted object by Danil Chapovalov · 5 years ago
  13. 3c5f91b Roll chromium_revision e74d6b592b..82de2e611e (698711:698813) by chromium-webrtc-autoroll · 5 years ago
  14. 289f313 Roll chromium_revision 5cbf4ebd59..e74d6b592b (698593:698711) by chromium-webrtc-autoroll · 5 years ago
  15. 4854b9f Roll chromium_revision 230cc8f7e4..5cbf4ebd59 (698466:698593) by chromium-webrtc-autoroll · 5 years ago
  16. 37ad5ab Change DatagramTransportInterface methods to pure virtual. by Bjorn A Mellem · 5 years ago
  17. 88db835 Change DataChannelTransportInterface/Sink methods to pure virtual. by Bjorn A Mellem · 5 years ago
  18. d702231 Cleanup deprecated monitoring of MediaTransport state. by Bjorn A Mellem · 5 years ago
  19. 5ac329c Cap h264 fuzzer input to 200k. by Patrik Höglund · 5 years ago
  20. 03bbef5 Fix accidental change of transport time metric by Johannes Kron · 5 years ago
  21. c2e9d84 Roll chromium_revision 303c57cf17..230cc8f7e4 (698351:698466) by chromium-webrtc-autoroll · 5 years ago
  22. 27b0e0d Remove obsolete todo comment in simulcast.h by Åsa Persson · 5 years ago
  23. 544dfb5 Delete isac GetBandwidthInfo/SetBandwidthInfo by Niels Möller · 5 years ago
  24. ef83cc5 Add fuzzer testing for Dependency Descriptor rtp header extension by Danil Chapovalov · 5 years ago
  25. 04fd215 Cleanup passing rtp packet to ulpfec receiver. by Danil Chapovalov · 5 years ago
  26. 0cff4fc Removed unused frame_size param from RtpFrameObject ctor. by philipel · 5 years ago
  27. 48b32b7 Delete support for enabling adaptive isac mode by Niels Möller · 5 years ago
  28. b5e4785 RtpFrameObject now takes an EncodedImageBuffer in its ctor. by philipel · 5 years ago
  29. fb59a6a Return `const char*` from ToString(RTCErrorType error). by Mirko Bonadei · 5 years ago
  30. e0b3167 Delete dead code inside #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED by Niels Möller · 5 years ago
  31. feee1e4 Add flag to APM to force multichannel even with AEC3 by Sam Zackrisson · 5 years ago
  32. e24557f Declare api:libjingle_peerconnection_api dependency on media:media_base by Niels Möller · 5 years ago
  33. 2051b8b Roll chromium_revision a536fa4a4a..303c57cf17 (698214:698351) by chromium-webrtc-autoroll · 5 years ago
  34. 95c538f Roll chromium_revision fc1e948f93..a536fa4a4a (698112:698214) by chromium-webrtc-autoroll · 5 years ago
  35. f288c8e Roll chromium_revision cf1a2beb4b..fc1e948f93 (697976:698112) by chromium-webrtc-autoroll · 5 years ago
  36. c12db81 Add frame receive to frame rendered metric to video_quality_analyzer by Johannes Kron · 5 years ago
  37. f0be5b5 Make GetBitstream non-virtual since it is no longer needed for testing. by philipel · 5 years ago
  38. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  39. ac315b2 Add support for max/min encode bitrate to peer connection quality test by Johannes Kron · 5 years ago
  40. 6a09263 Delete obsolete isac "assign" api by Niels Möller · 5 years ago
  41. d8ffbb0 Roll chromium_revision afdb2e7a8b..cf1a2beb4b (697871:697976) by chromium-webrtc-autoroll · 5 years ago
  42. 76161f7 Move the call to GetBitstream out of the RtpFrameObject ctor. by philipel · 5 years ago
  43. 14137a1 Adds logging of audio sessions status on the recording side in ADM for Android. by henrika · 5 years ago
  44. 86873f0 Improve field trial error message. by Björn Terelius · 5 years ago
  45. e942b14 New build target api:media_interface by Niels Möller · 5 years ago
  46. 0a5ed89 Adds remote estimates to rtc event log. by Sebastian Jansson · 5 years ago
  47. 6ed60e3 Implement Dependency Descriptor writer by Danil Chapovalov · 5 years ago
  48. 489843f Improve trendline estimator logging. by Björn Terelius · 5 years ago
  49. 693bf1e Delete modules/rtp_rtcp local DivideRoundToNearest in favor on one in rtc_base by Danil Chapovalov · 5 years ago
  50. bd24260 Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871) by chromium-webrtc-autoroll · 5 years ago
  51. efa04ef Roll chromium_revision 65274319fc..eae7ecf757 (697640:697744) by chromium-webrtc-autoroll · 5 years ago
  52. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  53. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
  54. 4bad650 Roll chromium_revision 2bd75c72c1..65274319fc (697505:697640) by chromium-webrtc-autoroll · 5 years ago
  55. 7b04a91 Delete almost all default methods on PeerConnectionInterface by Niels Möller · 5 years ago
  56. e607a06 Removed unused include from PacketBuffer. by philipel · 5 years ago
  57. 33b83fd Introduce integer division helpers with non-default rounding by Danil Chapovalov · 5 years ago
  58. b6a45dd Revert "Fix minor regression caused by a8336d3" by Evan Shrubsole · 5 years ago
  59. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
  60. 1162ba2 Add max/min encode bitrates to video config of peer connection tests by Johannes Kron · 5 years ago
  61. 7cfde54 Roll chromium_revision 51a0808947..2bd75c72c1 (697405:697505) by chromium-webrtc-autoroll · 5 years ago
  62. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 5 years ago
  63. c128df1 Update style guide for absl::make_unique. by Mirko Bonadei · 5 years ago
  64. 95c4b91 Roll chromium_revision 31d9542abc..51a0808947 (697288:697405) by chromium-webrtc-autoroll · 5 years ago
  65. ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 5 years ago
  66. 4d461ba Reusing MediaStreamAllocationConfig struct in ObserverConfig. by Sebastian Jansson · 5 years ago
  67. 86314cf Cleaning up C++14 move into lambda TODOs. by Sebastian Jansson · 5 years ago
  68. 368d002 Roll chromium_revision dbd1569418..31d9542abc (697157:697288) by chromium-webrtc-autoroll · 5 years ago
  69. 9fa8ef1 absl::make_unique presubmit check. by Mirko Bonadei · 5 years ago
  70. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  71. 809198e Fix minor regression caused by a8336d3 by Evan Shrubsole · 5 years ago
  72. 7d00342 Remove old packet socket factory header. by Patrik Höglund · 5 years ago
  73. e1b7777 Removing deprecated min_pacing_rate alias in StreamsConfig. by Sebastian Jansson · 5 years ago
  74. 4a822f4 Roll chromium_revision 2e4ccff8a8..dbd1569418 (696956:697157) by chromium-webrtc-autoroll · 5 years ago
  75. 2c6ea52 In TaskQueueTest::PostDelayedAfterDesctruct increase timeout by Danil Chapovalov · 5 years ago
  76. c1c6284 New (empty) build target api:media_stream_interface by Niels Möller · 5 years ago
  77. 1722182 Roll chromium_revision 3cf04dec00..2e4ccff8a8 (696812:696956) by chromium-webrtc-autoroll · 5 years ago
  78. 7262fc2 Refactor Rtp Receivers to accept SSRC 0. by Saurav Das · 5 years ago
  79. 3d16474 in RtcpTransciever use lambdas with move capture. by Danil Chapovalov · 5 years ago
  80. 3462793 Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812) by chromium-webrtc-autoroll · 5 years ago
  81. 68ef259 Delete deprecated rtc_event.h file by Danil Chapovalov · 5 years ago
  82. f5dec1c Implement Dependency Descriptor reader by Danil Chapovalov · 5 years ago
  83. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 5 years ago
  84. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 5 years ago
  85. 1b6a30d Update WebRTC's C++ style guide to reflect the switch to C++14. by Mirko Bonadei · 5 years ago
  86. a740142 Refactor LossNotificationController to not use VCMPacket by Niels Möller · 5 years ago
  87. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
  88. c4e80ad Delete forward declarations from peer_connection_interface.h by Niels Möller · 5 years ago
  89. 7af1bb3 Roll chromium_revision 9f15168729..1d12ff693d (696593:696696) by chromium-webrtc-autoroll · 5 years ago
  90. fcbe407 Adding more refined control over choice of band-splitting by Per Åhgren · 5 years ago
  91. ec06ebd Roll chromium_revision 9004bcf36a..9f15168729 (696490:696593) by chromium-webrtc-autoroll · 5 years ago
  92. 0dd37ce Roll chromium_revision 4740202690..9004bcf36a (696373:696490) by chromium-webrtc-autoroll · 5 years ago
  93. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  94. 6a6eb61 Roll chromium_revision f7cd88eb51..4740202690 (696270:696373) by chromium-webrtc-autoroll · 5 years ago
  95. e78fd80 New class DummyPeerConnection by Niels Möller · 5 years ago
  96. 3873927 Fix time units in plotted charts by Artem Titov · 5 years ago
  97. 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 5 years ago
  98. 0a7d5d8 Set console window NOTOPMOST flag after WindowFinderTest.FindDrawerWindow on Windows by Kimmo Kinnunen · 5 years ago
  99. 01be33b Using lambdas instead of rtc::Bind in BaseChannel. by Sebastian Jansson · 5 years ago
  100. 262bbae Fix rare audioLevel flake in RTCStatsIntegrationTest. by Henrik Boström · 5 years ago