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gerrit-public.fairphone.software / platform / external / webrtc / 52f7ae7c8988bcd1a0e4fad3dd80925a2ad482f5
  1. 52f7ae7 Make NetworkStateEstimator injectable in RemoteBitrateEstimator by Per Kjellander · 4 years, 11 months ago
  2. 467073a Revert "Adds peer scenario connection interface." by Qingsi Wang · 4 years, 11 months ago
  3. 437077d Revert "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" by Qingsi Wang · 4 years, 11 months ago
  4. f6aa572 First step for introducing multichannel support for the AEC3 capture by Per Åhgren · 4 years, 11 months ago
  5. 2dc1425 Roll chromium_revision a87779d34b..56140e7d8b (695071:695187) by chromium-webrtc-autoroll · 4 years, 11 months ago
  6. de5f639 Removes decoder thread fallback from VideoReceiveStream. by Sebastian Jansson · 4 years, 11 months ago
  7. 29ab487 Revert "Removes string support in field trial parser." by Philip Eliasson · 4 years, 11 months ago
  8. 507f434 Reland "Make relative arrival delay mode default in NetEq delay manager." by Jakob Ivarsson · 5 years ago
  9. 3354157 Add support for 192kHz input audio sample rate. by henrika · 4 years, 11 months ago
  10. 45b01c7 Delete some dead code in vcm::VideoReceiver and VCMReceiver by Niels Möller · 4 years, 11 months ago
  11. fe407b7 Move code related to VideoCodingModule to its own build target by Niels Möller · 4 years, 11 months ago
  12. 01b7e92 Mark test::DriftingClock constants as constexpr by Danil Chapovalov · 4 years, 11 months ago
  13. 2486aeb Add ability to disable PSNR and SSIM computation in DVQA by Artem Titov · 4 years, 11 months ago
  14. f7b1aa4 Fixing some typos. by Mirko Bonadei · 4 years, 11 months ago
  15. 01e97ae Move docs about native code development into a repo directory. by Mirko Bonadei · 5 years ago
  16. 56d89da Roll chromium_revision e25e764221..a87779d34b (694813:695071) by chromium-webrtc-autoroll · 4 years, 11 months ago
  17. 9bc9885 Add placeholder target to move rtc_error out of the main API target. by Mirko Bonadei · 4 years, 11 months ago
  18. ee84d39 AEC3: Downmix multichannel signals before delay estimation by Gustaf Ullberg · 4 years, 11 months ago
  19. d181ee7 Adds peer scenario connection interface. by Sebastian Jansson · 4 years, 11 months ago
  20. 0cd61b6 MultiCodecReceiveTest: fix for flaky test. by Åsa Persson · 4 years, 11 months ago
  21. b3f1487 Add ability to provide TEXT hint only when requested in PC framework by Artem Titov · 4 years, 11 months ago
  22. 9509d95 Add empty build target modules/video_coding:video_coding_legacy by Niels Möller · 4 years, 11 months ago
  23. c77df78 Revert "Improve performance of RtpPacketHistory" by Qingsi Wang · 4 years, 11 months ago
  24. 487f9a1 Reland "Refactor SCTP data channels to use DataChannelTransportInterface." by Bjorn A Mellem · 4 years, 11 months ago
  25. 116ffe7 Switch to compiling WebRTC -std=c++14 by default by Danil Chapovalov · 5 years ago
  26. a0e6ded Roll chromium_revision 75cf3925c2..e25e764221 (694706:694813) by chromium-webrtc-autoroll · 4 years, 11 months ago
  27. 9e380fd Improve performance of RtpPacketHistory by Erik Språng · 4 years, 11 months ago
  28. a5d952f Reland "Refactor FEC code to use COW buffers" by Ilya Nikolaevskiy · 5 years ago
  29. 4d7dac6 Remove usage of RtpRtcp::SetSSRC() in RtpRtcpImplTest by Erik Språng · 4 years, 11 months ago
  30. 0987273 Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video. by Andrei Dumitru · 5 years ago
  31. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 4 years, 11 months ago
  32. a8336d3 Connect the stable target rate to the video encoders by Florent Castelli · 4 years, 11 months ago
  33. ddef8d1 Add support of displaying video during the PC level test by Artem Titov · 5 years ago
  34. 2aaf66e Roll chromium_revision 82910f739a..75cf3925c2 (694601:694706) by chromium-webrtc-autoroll · 4 years, 11 months ago
  35. f7cb16f Check input parameter in RemoteEstimatorProxy::IncomingPacket without lock by Per Kjellander · 4 years, 11 months ago
  36. ee3d995 New class VideoReceiver2, a trimmed-down vcm::VideoReceiver by Niels Möller · 4 years, 11 months ago
  37. 4d6b269 Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM by henrika · 5 years ago
  38. 81a08a7 Roll chromium_revision b67ee864bb..82910f739a (694439:694601) by chromium-webrtc-autoroll · 5 years ago
  39. be2e5f7 Make payload type demux conditional on media direction by Steve Anton · 5 years ago
  40. 3146ea0 Roll chromium_revision 50f07bc317..b67ee864bb (694314:694439) by chromium-webrtc-autoroll · 5 years ago
  41. 20232a9 Use obfuscated IPs in logging in p2p/ and pc/. by Qingsi Wang · 5 years ago
  42. 25a4b06 Roll chromium_revision df72a9a5c5..50f07bc317 (694187:694314) by chromium-webrtc-autoroll · 5 years ago
  43. b64d65e Fix NetworkEmulationManagerTest.ThroughputStats flakiness. by Yves Gerey · 5 years ago
  44. e611f19 Remove completed TODOs by Sam Zackrisson · 5 years ago
  45. 30323e2 VP9 screenshare: fix incorrect assumptions on buffer contents by Ilya Nikolaevskiy · 5 years ago
  46. e15c10a Fix for rare read of uninitialized value in remote estimate test. by Sebastian Jansson · 5 years ago
  47. 2b9dba3 Implement stable rate support in SimulcastRateAllocator by Erik Språng · 5 years ago
  48. 059a0b7 Fix for deadlock in AudioUsesAbsSendTimeExtension test. by Sebastian Jansson · 5 years ago
  49. a3baf2a Add one more BasicPortAllocator constructor by Niels Möller · 5 years ago
  50. cc7ea18 Roll chromium_revision 3cccc3ec06..df72a9a5c5 (694083:694187) by chromium-webrtc-autoroll · 5 years ago
  51. 19f9c2a Refactor video analyzer injection helper by Artem Titov · 5 years ago
  52. 0bd2eff Reland "New build target p2p:stun_types" by Niels Möller · 5 years ago
  53. 662e31f Prepare to move packet_socket_factory to api/. by Patrik Höglund · 5 years ago
  54. 50b0baf [tsan] Guard TestStatictics against concurrent access. by Yves Gerey · 5 years ago
  55. 75e2290 Rollback to strict audio codec tests for libopus on android (neon). by Yves Gerey · 5 years ago
  56. ba5f8e9 Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5." by Guido Urdaneta · 5 years ago
  57. c2accf2 [tsan] Guard OwnerThread against concurrent access. by Yves Gerey · 5 years ago
  58. eea6ced Roll chromium_revision 675968a8c6..3cccc3ec06 (693954:694083) by chromium-webrtc-autoroll · 5 years ago
  59. 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 5 years ago
  60. f6ec68d Roll chromium_revision 7e5c36432b..675968a8c6 (693630:693954) by chromium-webrtc-autoroll · 5 years ago
  61. e715301 Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5. by Trevor Hayes · 5 years ago
  62. bbbae42 Refactor video analyzer injection helper by Artem Titov · 5 years ago
  63. d4e6904 AEC3: Reducing the complexity and heap usage of the adaptive filter by Per Åhgren · 5 years ago
  64. f294d26 Delete deprecated method StreamStatistician::GetStatistics by Niels Möller · 5 years ago
  65. cfb9497 Add multi-channel to FftBuffer by Sam Zackrisson · 5 years ago
  66. 5b728cc Revert "Make relative arrival delay mode default in NetEq delay manager." by Alessio Bazzica · 5 years ago
  67. 8dcbdd2 Roll chromium_revision e96090c328..7e5c36432b (693514:693630) by chromium-webrtc-autoroll · 5 years ago
  68. f3f6159 Rename VectorBuffer->SpectrumBuffer, MatrixBuffer->BlockBuffer, BlockBuffer->Aec2BlockBuffer by Sam Zackrisson · 5 years ago
  69. 77c71d1 Make relative arrival delay mode default in NetEq delay manager. by Jakob Ivarsson · 5 years ago
  70. a81c09d Make VectorBuffer in AEC3 multi-channel by Sam Zackrisson · 5 years ago
  71. 9305d11 Delete deprecated rtc_event_log_factory_interface.h by Danil Chapovalov · 5 years ago
  72. 24b945d Add support of AudioRecord.Builder in the ADM for Android by henrika · 5 years ago
  73. 065dd27 Roll chromium_revision c27b8dde0e..e96090c328 (693394:693514) by chromium-webrtc-autoroll · 5 years ago
  74. e333505 Roll chromium_revision b931c7fd8b..c27b8dde0e (693252:693394) by chromium-webrtc-autoroll · 5 years ago
  75. bbca6dd Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. by Mirko Bonadei · 5 years ago
  76. c51b4e3 Roll chromium_revision f661d57809..b931c7fd8b (693124:693252) by chromium-webrtc-autoroll · 5 years ago
  77. 09dcafd Revert "Always create output_dir in setup_apprtc.py." by Mirko Bonadei · 5 years ago
  78. cf9cbf5 Add support for stable bitrate target in SvcRateAllocator by Erik Språng · 5 years ago
  79. 1067d31 Make the stable target rate always less or equal than the target rate by Florent Castelli · 5 years ago
  80. 52a8da3 Always create output_dir in setup_apprtc.py. by Mirko Bonadei · 5 years ago
  81. d9f98cd Roll chromium_revision 248662b1b8..f661d57809 (693000:693124) by chromium-webrtc-autoroll · 5 years ago
  82. 77d197f Make video_capture_internal_impl publicly visible. by Mirko Bonadei · 5 years ago
  83. e74156f Removes string support in field trial parser. by Sebastian Jansson · 5 years ago
  84. 3247244 Delete unused method AudioCodingModule::GetDecodingCallStatistics by Niels Möller · 5 years ago
  85. a33dc01 AEC3: Propagate the number of channels to the adaptive filters by Per Åhgren · 5 years ago
  86. 1a3859c Simplify book-keeping of lost packets by Niels Möller · 5 years ago
  87. 340e0c5 Delete old version of PeerConnection::SetConfiguration by Niels Möller · 5 years ago
  88. 59e1464 Fix 28 ClangTidy - Readability findings in modules/rtp_rtcp/ by Danil Chapovalov · 5 years ago
  89. 38350b1 Roll chromium_revision d74690feb1..248662b1b8 (692875:693000) by chromium-webrtc-autoroll · 5 years ago
  90. e007ad1 Roll chromium_revision 5e84fd2515..d74690feb1 (692730:692875) by chromium-webrtc-autoroll · 5 years ago
  91. da10032 Roll chromium_revision da46a51bc2..5e84fd2515 (692597:692730) by chromium-webrtc-autoroll · 5 years ago
  92. 7cdcda9 Use the sanitized pair when surfacing the candidate pair change event. by Qingsi Wang · 5 years ago
  93. 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 5 years ago
  94. 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 5 years ago
  95. 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 5 years ago
  96. 23003a2 Add saza to audio watchlists by Sam Zackrisson · 5 years ago
  97. b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 5 years ago
  98. 1aa7e2f Roll chromium_revision 8304ddd943..da46a51bc2 (692489:692597) by chromium-webrtc-autoroll · 5 years ago
  99. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  100. f2773b5 Add webrtc_apprtc_browsertest.cc resource dir to .gitignore. by Mirko Bonadei · 5 years ago