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gerrit-public.fairphone.software / platform / external / webrtc / a06048a41ed1088b490671007c151b68f483eeef
  1. a06048a Return status instead of CHECKing in event log parser. by Björn Terelius · 5 years ago
  2. cc9bf63 Revert "Correct AEC3 multichannel functionality activation" by Per Åhgren · 5 years ago
  3. 191e38f Delete gturn support by Niels Möller · 5 years ago
  4. 9dda1b3 Correct AEC3 multichannel functionality activation by Per Åhgren · 5 years ago
  5. 8846c8a RNN VAD: cast and scale quantized weights at init by Alessio Bazzica · 5 years ago
  6. 26452ff Cleanup of TransportFeedbackAdapter. by Sebastian Jansson · 5 years ago
  7. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 5 years ago
  8. 2ebbff8 do not offer gcm as the preferred cipher suite by Philipp Hancke · 5 years ago
  9. 3ce44a3 Move NetEq headers to api/ by Ivo Creusen · 5 years ago
  10. 739a5b3 Refactors BitrateProber with unit types and absolute probe time. by Erik Språng · 5 years ago
  11. 8d65e9a Fixes pacing interval dependency and race in BandwidthEndToEndTest by Erik Språng · 5 years ago
  12. caaa9e7 AEC3: Handle multichannel audio in single CNG instance by Gustaf Ullberg · 5 years ago
  13. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
  14. d1ea4c9 Update comments on Audio Level RTP header extension. by Minyue Li · 5 years ago
  15. 577c580 Do not stop SingleThreadedTaskQueueForTestingTest near the end of the tests by Danil Chapovalov · 5 years ago
  16. fb075d5 Removing unused Opus wrapper API: WebRTCOpus_DecodePlc. by Minyue Li · 5 years ago
  17. 0cbb58e Reland "Refactoring of the noise suppressor and adding true multichannel support" by Per Åhgren · 5 years ago
  18. 159b417 Keep the video send stream alive if the encoder drop frames. by Jakob Ivarsson · 5 years ago
  19. c35333d Add RTC_EXPORT_TEMPLATE_{DECLARE,DEFINE} macros. by Mirko Bonadei · 5 years ago
  20. de36595 Added new Apple devices. by Yura Yaroshevich · 5 years ago
  21. 32913c1 Removes the flakiness in PeerConnectionUsageHistogramTest. by Qingsi Wang · 5 years ago
  22. 5bd8cb7 Revert "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."" by Henrik Boström · 5 years ago
  23. 54d0278 Renaming opus_interface.c to opus_interface.cc. by Minyue Li · 5 years ago
  24. 09860e0 Split out counting unique rtp timestamps from packet_buffer by Danil Chapovalov · 5 years ago
  25. a0adf3d Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5." by Trevor Hayes · 5 years ago
  26. 9560d7d Make update_rect optional in VideoFrame by Ilya Nikolaevskiy · 5 years ago
  27. 4e19670 [PeerConnection] Implement parameterless SetLocalDescription(). by Henrik Boström · 5 years ago
  28. 9b66114 Disable rendering statistics while video is paused. by Sami Kalliomäki · 5 years ago
  29. 4778f6c Revert "Refactoring of the noise suppressor and adding true multichannel support" by Artem Titov · 5 years ago
  30. 9c712bb Fix invalid @Nullable handling in TextureBufferImpl. by Sami Kalliomäki · 5 years ago
  31. f298855 Cleanup of feedback observer interface by Sebastian Jansson · 5 years ago
  32. 470b2d5 Stop relying on GN's sources_assignment_filter. by Mirko Bonadei · 5 years ago
  33. ae40e19 AEC3: Adding a configurable render signal gain by Per Åhgren · 5 years ago
  34. 87a7b82 Refactoring of the noise suppressor and adding true multichannel support by Per Åhgren · 5 years ago
  35. c6c3f86 Expose TLS version and SRTP cipher to API by Harald Alvestrand · 5 years ago
  36. 6981fb5 Add support to not use turn server as stun server. by Honghai Zhang · 5 years ago
  37. 74f96ec Removes unused late feedback plot from analyzer. by Sebastian Jansson · 5 years ago
  38. 9cdc9cc Cleanup of deprecated RTPSender code by Erik Språng · 5 years ago
  39. cb30726 Remove deprecated Audio Processing APIs by Gustaf Ullberg · 5 years ago
  40. 6e4e688 Fixed MSAN issue with usrsctp reliability test. by Yura Yaroshevich · 5 years ago
  41. fbec2ec Detach H264 sps pps tracker from VCMPacket by Danil Chapovalov · 5 years ago
  42. 05c4792 Removes OnPacketAdded callback from feedback adapter. by Sebastian Jansson · 5 years ago
  43. 9c71e49 Remove redundant BitrateProber::OnIncomingPacket() call by Erik Språng · 5 years ago
  44. 01a21f7 Roll chromium_revision 9109135db0..7ce0264138 (709913:710014) by chromium-webrtc-autoroll · 5 years ago
  45. 77b7529 Reland "Use RtpSenderEgress directly instead of via RTPSender" by Erik Språng · 5 years ago
  46. 79e653c Apply bitrate boosting depending on field-trial. by Ivo Creusen · 5 years ago
  47. 6f5b9e0 Roll chromium_revision d68d92fb45..9109135db0 (709806:709913) by chromium-webrtc-autoroll · 5 years ago
  48. 70770ac Make AudioFrame member instead of raw pointer in APM test fixture by Sam Zackrisson · 5 years ago
  49. cff20c2 Adds protected bitrate helper methods to RtpRtcpImpl by Erik Språng · 5 years ago
  50. a3728d3 Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." by Henrik Boström · 5 years ago
  51. f50d58b Add .clangd to .gitignore by Rasmus Brandt · 5 years ago
  52. 5cb7807 Implement crypto stats on DTLS transport by Harald Alvestrand · 5 years ago
  53. a81e2b4 Revert "Use RtpSenderEgress directly instead of via RTPSender" by Erik Språng · 5 years ago
  54. b533010 Use RtpSenderEgress directly instead of via RTPSender by Erik Språng · 5 years ago
  55. 3eae7e4 Add exponential backoff of retransmissions for a given packet by Erik Språng · 5 years ago
  56. 632d57d Ignore low probe results when using NetworkStateEstimator under field trial by Per Kjellander · 5 years ago
  57. 1230fb7 ICE : add field trial for initial select dampening by Jonas Oreland · 5 years ago
  58. c189ace Roll chromium_revision 04c3c4c8f1..d68d92fb45 (709704:709806) by chromium-webrtc-autoroll · 5 years ago
  59. e38e119 Roll chromium_revision 98ef1d6866..04c3c4c8f1 (709549:709704) by chromium-webrtc-autoroll · 5 years ago
  60. e95fc85 Roll chromium_revision 3c5165bebc..98ef1d6866 (709394:709549) by chromium-webrtc-autoroll · 5 years ago
  61. 91e3ebe Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."" by Mirko Bonadei · 5 years ago
  62. 29db239 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  63. e114fb6 Added usrsctp reliablitiy stress test. by Yura Yaroshevich · 5 years ago
  64. 67ac9e8 Prepares RTPSender for extracting RtpSenderEgress by Erik Språng · 5 years ago
  65. 492fdf4 Make rtc_json poisonous in WebRTC by Sam Zackrisson · 5 years ago
  66. 1a61739 Fix MemoryLogWriter so that it always writes the full data. by Björn Terelius · 5 years ago
  67. 53a31f7 Introduce injectable NetEqController interface. by Ivo Creusen · 5 years ago
  68. 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 5 years ago
  69. 49c0880 Revert "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." by Henrik Boström · 5 years ago
  70. 1dac707 Cleanup PacketBuffer tests to use immediate result by Danil Chapovalov · 5 years ago
  71. 4f2783b Speculative Revert: "Use FakeRenderer when fuzzing" by Patrik Höglund · 5 years ago
  72. b394a56 Cleanup of EchoControl interface after downstream fixes by Gustaf Ullberg · 5 years ago
  73. e277bde Add APM test of pre-amplifier gain by Sam Zackrisson · 5 years ago
  74. e914c1e Roll chromium_revision 64883b3ea2..3c5165bebc (709283:709394) by chromium-webrtc-autoroll · 5 years ago
  75. 1dddaa1 [PeerConnection] Use an OperationsChain in PeerConnection for async ops. by Henrik Boström · 5 years ago
  76. ef0e4d0 Roll chromium_revision e1ab9e9b20..64883b3ea2 (709180:709283) by chromium-webrtc-autoroll · 5 years ago
  77. c4af214 Roll chromium_revision d7338c33b2..e1ab9e9b20 (708965:709180) by chromium-webrtc-autoroll · 5 years ago
  78. 3cb6104 AEC3: Support negative delay with external delay estimator by Gustaf Ullberg · 5 years ago
  79. a922904 Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly. by Erik Språng · 5 years ago
  80. b2290f4 Revert "Reset end-of-frame flag in non-VCL packet." by Sergey Silkin · 5 years ago
  81. fc78aac Batches video frame packets when posting to pacer by Erik Språng · 5 years ago
  82. 2040dcf Roll chromium_revision f656c810e4..d7338c33b2 (708845:708965) by chromium-webrtc-autoroll · 5 years ago
  83. eec3919 Remove trial WebRTC-Bwe-ProbeRateFallback by Per Kjellander · 5 years ago
  84. d113ee3 Removes deprecated WebRTC-Bwe-AimdRateControl-NetworkState trial. by Sebastian Jansson · 5 years ago
  85. bd20077 Roll chromium_revision 510c0ca3d7..f656c810e4 (708742:708845) by chromium-webrtc-autoroll · 5 years ago
  86. 0e2b581 RTC_EXPORT webrtc::DesktopCapturerDifferWrapper. by Mirko Bonadei · 5 years ago
  87. c1a8abc Roll chromium_revision b5030084da..510c0ca3d7 (708640:708742) by chromium-webrtc-autoroll · 5 years ago
  88. 4ff1c87 Fix RTC_LOCKABLE RTC_EXPORT order for rtc::Thread. by Mirko Bonadei · 5 years ago
  89. d7bf5c5 Revert "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  90. 6e81567 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  91. ce1ffcd change PacketBuffer to return it's result rather that use callback by Danil Chapovalov · 5 years ago
  92. 2522b25 Roll chromium_revision 6dc3a51e22..b5030084da (708537:708640) by chromium-webrtc-autoroll · 5 years ago
  93. 6adb0a2 Do not compile webrtc_lib_link_test if is_asan=true. by Mirko Bonadei · 5 years ago
  94. 21bfa40 Update APM config on RuntimeSetting pre amplifier gain change by Sam Zackrisson · 5 years ago
  95. 4f178d0 Fix gtk color-space conversion in peerconnection_client by Niels Möller · 5 years ago
  96. d81a04e Roll chromium_revision c0cca6e419..6dc3a51e22 (708426:708537) by chromium-webrtc-autoroll · 5 years ago
  97. 0ff7c02 Add multipleTouchEnabled for subview of RTCMTLVideoView and RTCEAGLVideoView by CZ Theng · 5 years ago
  98. 27c2936 Implement an OperationsChain, to be used by PeerConnection in follow-up. by Henrik Boström · 5 years ago
  99. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  100. 261fc51 Roll chromium_revision 83bb172f2d..c0cca6e419 (708314:708426) by chromium-webrtc-autoroll · 5 years ago