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gerrit-public.fairphone.software / platform / external / webrtc / 91c824f8494ee15a821de7e58e899c14bb459a6c
  1. 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 5 years ago
  2. f6ec68d Roll chromium_revision 7e5c36432b..675968a8c6 (693630:693954) by chromium-webrtc-autoroll · 5 years ago
  3. 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
  4. bbbae42 Refactor video analyzer injection helper by Artem Titov · 5 years ago
  5. d4e6904 AEC3: Reducing the complexity and heap usage of the adaptive filter by Per Åhgren · 5 years ago
  6. f294d26 Delete deprecated method StreamStatistician::GetStatistics by Niels Möller · 5 years ago
  7. cfb9497 Add multi-channel to FftBuffer by Sam Zackrisson · 5 years ago
  8. 5b728cc Revert "Make relative arrival delay mode default in NetEq delay manager." by Alessio Bazzica · 5 years ago
  9. 8dcbdd2 Roll chromium_revision e96090c328..7e5c36432b (693514:693630) by chromium-webrtc-autoroll · 5 years ago
  10. f3f6159 Rename VectorBuffer->SpectrumBuffer, MatrixBuffer->BlockBuffer, BlockBuffer->Aec2BlockBuffer by Sam Zackrisson · 5 years ago
  11. 77c71d1 Make relative arrival delay mode default in NetEq delay manager. by Jakob Ivarsson · 5 years ago
  12. a81c09d Make VectorBuffer in AEC3 multi-channel by Sam Zackrisson · 5 years ago
  13. 9305d11 Delete deprecated rtc_event_log_factory_interface.h by Danil Chapovalov · 5 years ago
  14. 24b945d Add support of AudioRecord.Builder in the ADM for Android by henrika · 5 years ago
  15. 065dd27 Roll chromium_revision c27b8dde0e..e96090c328 (693394:693514) by chromium-webrtc-autoroll · 5 years ago
  16. e333505 Roll chromium_revision b931c7fd8b..c27b8dde0e (693252:693394) by chromium-webrtc-autoroll · 5 years ago
  17. bbca6dd Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. by Mirko Bonadei · 5 years ago
  18. c51b4e3 Roll chromium_revision f661d57809..b931c7fd8b (693124:693252) by chromium-webrtc-autoroll · 5 years ago
  19. 09dcafd Revert "Always create output_dir in setup_apprtc.py." by Mirko Bonadei · 5 years ago
  20. cf9cbf5 Add support for stable bitrate target in SvcRateAllocator by Erik Språng · 5 years ago
  21. 1067d31 Make the stable target rate always less or equal than the target rate by Florent Castelli · 5 years ago
  22. 52a8da3 Always create output_dir in setup_apprtc.py. by Mirko Bonadei · 5 years ago
  23. d9f98cd Roll chromium_revision 248662b1b8..f661d57809 (693000:693124) by chromium-webrtc-autoroll · 5 years ago
  24. 77d197f Make video_capture_internal_impl publicly visible. by Mirko Bonadei · 5 years ago
  25. e74156f Removes string support in field trial parser. by Sebastian Jansson · 5 years ago
  26. 3247244 Delete unused method AudioCodingModule::GetDecodingCallStatistics by Niels Möller · 5 years ago
  27. a33dc01 AEC3: Propagate the number of channels to the adaptive filters by Per Åhgren · 5 years ago
  28. 1a3859c Simplify book-keeping of lost packets by Niels Möller · 5 years ago
  29. 340e0c5 Delete old version of PeerConnection::SetConfiguration by Niels Möller · 5 years ago
  30. 59e1464 Fix 28 ClangTidy - Readability findings in modules/rtp_rtcp/ by Danil Chapovalov · 5 years ago
  31. 38350b1 Roll chromium_revision d74690feb1..248662b1b8 (692875:693000) by chromium-webrtc-autoroll · 5 years ago
  32. e007ad1 Roll chromium_revision 5e84fd2515..d74690feb1 (692730:692875) by chromium-webrtc-autoroll · 5 years ago
  33. da10032 Roll chromium_revision da46a51bc2..5e84fd2515 (692597:692730) by chromium-webrtc-autoroll · 5 years ago
  34. 7cdcda9 Use the sanitized pair when surfacing the candidate pair change event. by Qingsi Wang · 5 years ago
  35. 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 5 years ago
  36. 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 5 years ago
  37. 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 5 years ago
  38. 23003a2 Add saza to audio watchlists by Sam Zackrisson · 5 years ago
  39. b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 5 years ago
  40. 1aa7e2f Roll chromium_revision 8304ddd943..da46a51bc2 (692489:692597) by chromium-webrtc-autoroll · 5 years ago
  41. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  42. f2773b5 Add webrtc_apprtc_browsertest.cc resource dir to .gitignore. by Mirko Bonadei · 5 years ago
  43. 082696e Revert "Refactor FEC code to use COW buffers" by Ilya Nikolaevskiy · 5 years ago
  44. ce202a0 Reland "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."" by Per Åhgren · 5 years ago
  45. a77a1f9 Roll chromium_revision 78591f12ff..8304ddd943 (692389:692489) by chromium-webrtc-autoroll · 5 years ago
  46. 5056af0 Make sure link allocation is at least as large as bitrate sum. by Erik Språng · 5 years ago
  47. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  48. d112c75 Revert "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."" by Per Åhgren · 5 years ago
  49. 65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 5 years ago
  50. 5b4fcb5 New build target p2p:stun_types by Niels Möller · 5 years ago
  51. eec5fff Refactor FEC code to use COW buffers by Ilya Nikolaevskiy · 5 years ago
  52. a66395e Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3." by Per Åhgren · 5 years ago
  53. 8b7c5e4 Add empty build target p2p:stun_types by Niels Möller · 5 years ago
  54. 54c0326 Roll chromium_revision a42eacf137..78591f12ff (692288:692389) by chromium-webrtc-autoroll · 5 years ago
  55. 602942f Filter out small packets from delay-based overuse detection. by Bjorn Terelius · 5 years ago
  56. f660e81 Revert "Simplify pacer queue" by Erik Språng · 5 years ago
  57. ce6a0c8 Roll chromium_revision 54ad211b04..a42eacf137 (692182:692288) by chromium-webrtc-autoroll · 5 years ago
  58. ed2fc50 Roll chromium_revision 291798b89f..54ad211b04 (692040:692182) by chromium-webrtc-autoroll · 5 years ago
  59. af3afff Roll chromium_revision ea980c903b..291798b89f (691937:692040) by chromium-webrtc-autoroll · 5 years ago
  60. a42b632 Adding CreateTcpClientSocket without user_agent and proxy_info. by Patrik Höglund · 5 years ago
  61. 8b14b0d Revert "Refactor SCTP data channels to use DataChannelTransportInterface." by Henrik Boström · 5 years ago
  62. 066b42f Interface for monitoring ref counts of texture buffers created by SurfaceTextureHelper. by Sami Kalliomäki · 5 years ago
  63. b6220d9 Delete unused logic for audio RtcpMode::kOff by Niels Möller · 5 years ago
  64. e3e30ae Revert "Add core multi-channel pipeline in AEC3" by Ilya Nikolaevskiy · 5 years ago
  65. ddd50ef Use HasOneRef to ensure safe reallocation of buffer in EncodedImage by Ilya Nikolaevskiy · 5 years ago
  66. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  67. 4894fde Fix test_support_unittests with enable_iterator_debugging=true by Kimmo Kinnunen · 5 years ago
  68. 9f00f0e Add support for unsigned parameters in FieldTrialParser by Bjorn Terelius · 5 years ago
  69. f3a197e Add core multi-channel pipeline in AEC3 by Per Åhgren · 5 years ago
  70. 01a4918 Roll chromium_revision 71facea151..ea980c903b (691823:691937) by chromium-webrtc-autoroll · 5 years ago
  71. 257ce72 Roll chromium_revision 3a1d849d09..71facea151 (691713:691823) by chromium-webrtc-autoroll · 5 years ago
  72. 3e8bf28 Increase the maximum supported sample rate to 384000 Hz and add tests by Per Åhgren · 5 years ago
  73. 8577729 Roll chromium_revision 52323b9fe0..3a1d849d09 (691589:691713) by chromium-webrtc-autoroll · 5 years ago
  74. 4c85828 Refactor SCTP data channels to use DataChannelTransportInterface. by Bjorn A Mellem · 5 years ago
  75. 55dd72c Remove lock for process thread pointer from PacedSender. by Tommi · 5 years ago
  76. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 5 years ago
  77. 022a7c8 Fix HexEncodeTest.TestZeroLengthNoDelimiter with enable_iterator_debugging=true by Kimmo Kinnunen · 5 years ago
  78. 8226875 Avoids race during VideoStreamEncoder unittest teardown by Erik Språng · 5 years ago
  79. 640aee2 Remove backwards compatibility names from api/uma_metrics.h. by Mirko Bonadei · 5 years ago
  80. da2f4a3 Remove stale TODO from rtc_base/checks.h. by Mirko Bonadei · 5 years ago
  81. 44dc241 Allows configuration of playout audio buffer by Alex Narest · 5 years ago
  82. 008213a Roll chromium_revision 9dd4f35a9d..52323b9fe0 (691474:691589) by chromium-webrtc-autoroll · 5 years ago
  83. 16946e3 Remove unused StorageType enum by Erik Språng · 5 years ago
  84. a2479f7 Remove minimum bucket returned by histogram quantile function. by Jakob Ivarsson · 5 years ago
  85. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  86. 3fa4938 Increased event log visualizer RTP clock estimation tolerance. by philipel · 5 years ago
  87. 7fa4277 Fix for tsan failue in real time scenario tests. by Sebastian Jansson · 5 years ago
  88. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  89. 7db900e Simplify pacer queue by Erik Språng · 5 years ago
  90. 228900f Add TURN_LOGGING_ID to android sdk by Jonas Oreland · 5 years ago
  91. 03307cb Roll chromium_revision acc73a9128..9dd4f35a9d (691348:691474) by chromium-webrtc-autoroll · 5 years ago
  92. 5245501 Roll chromium_revision 8538f0b743..acc73a9128 (691247:691348) by chromium-webrtc-autoroll · 5 years ago
  93. 703ea95 Only create a datagram RTP transport if the datagram transport should be used for RTP. by Bjorn A Mellem · 5 years ago
  94. dbec6d3 Roll chromium_revision f706cf738b..8538f0b743 (690793:691247) by chromium-webrtc-autoroll · 5 years ago
  95. fa046b3 Remove unused using statements in webrtc_sdp.cc by Elad Alon · 5 years ago
  96. 10b6361 Add license for android_ndk by Oleksandr Iakovenko · 5 years ago
  97. d191533 Fix wrong-import-order pylint errors in quality_assessment.signal_processing module. by Oleksandr Iakovenko · 5 years ago
  98. 4b9701e Fix simulcast tests and PC framework for conference mode support by Artem Titov · 5 years ago
  99. 149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 5 years ago
  100. 3b69817 Revert "Reland "Preserve min and max playout delay from RTP header extension"" by Johannes Kron · 5 years ago