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gerrit-public.fairphone.software / platform / external / webrtc / f298855981d7ce4dce0b8338ad5994db048ec0e1
  1. f298855 Cleanup of feedback observer interface by Sebastian Jansson · 5 years ago
  2. 470b2d5 Stop relying on GN's sources_assignment_filter. by Mirko Bonadei · 5 years ago
  3. ae40e19 AEC3: Adding a configurable render signal gain by Per Åhgren · 5 years ago
  4. 87a7b82 Refactoring of the noise suppressor and adding true multichannel support by Per Åhgren · 5 years ago
  5. c6c3f86 Expose TLS version and SRTP cipher to API by Harald Alvestrand · 5 years ago
  6. 6981fb5 Add support to not use turn server as stun server. by Honghai Zhang · 5 years ago
  7. 74f96ec Removes unused late feedback plot from analyzer. by Sebastian Jansson · 5 years ago
  8. 9cdc9cc Cleanup of deprecated RTPSender code by Erik Språng · 5 years ago
  9. cb30726 Remove deprecated Audio Processing APIs by Gustaf Ullberg · 5 years ago
  10. 6e4e688 Fixed MSAN issue with usrsctp reliability test. by Yura Yaroshevich · 5 years ago
  11. fbec2ec Detach H264 sps pps tracker from VCMPacket by Danil Chapovalov · 5 years ago
  12. 05c4792 Removes OnPacketAdded callback from feedback adapter. by Sebastian Jansson · 5 years ago
  13. 9c71e49 Remove redundant BitrateProber::OnIncomingPacket() call by Erik Språng · 5 years ago
  14. 01a21f7 Roll chromium_revision 9109135db0..7ce0264138 (709913:710014) by chromium-webrtc-autoroll · 5 years ago
  15. 77b7529 Reland "Use RtpSenderEgress directly instead of via RTPSender" by Erik Språng · 5 years ago
  16. 79e653c Apply bitrate boosting depending on field-trial. by Ivo Creusen · 5 years ago
  17. 6f5b9e0 Roll chromium_revision d68d92fb45..9109135db0 (709806:709913) by chromium-webrtc-autoroll · 5 years ago
  18. 70770ac Make AudioFrame member instead of raw pointer in APM test fixture by Sam Zackrisson · 5 years ago
  19. cff20c2 Adds protected bitrate helper methods to RtpRtcpImpl by Erik Språng · 5 years ago
  20. a3728d3 Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." by Henrik Boström · 5 years ago
  21. f50d58b Add .clangd to .gitignore by Rasmus Brandt · 5 years ago
  22. 5cb7807 Implement crypto stats on DTLS transport by Harald Alvestrand · 5 years ago
  23. a81e2b4 Revert "Use RtpSenderEgress directly instead of via RTPSender" by Erik Språng · 5 years ago
  24. b533010 Use RtpSenderEgress directly instead of via RTPSender by Erik Språng · 5 years ago
  25. 3eae7e4 Add exponential backoff of retransmissions for a given packet by Erik Språng · 5 years ago
  26. 632d57d Ignore low probe results when using NetworkStateEstimator under field trial by Per Kjellander · 5 years ago
  27. 1230fb7 ICE : add field trial for initial select dampening by Jonas Oreland · 5 years ago
  28. c189ace Roll chromium_revision 04c3c4c8f1..d68d92fb45 (709704:709806) by chromium-webrtc-autoroll · 5 years ago
  29. e38e119 Roll chromium_revision 98ef1d6866..04c3c4c8f1 (709549:709704) by chromium-webrtc-autoroll · 5 years ago
  30. e95fc85 Roll chromium_revision 3c5165bebc..98ef1d6866 (709394:709549) by chromium-webrtc-autoroll · 5 years ago
  31. 91e3ebe Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."" by Mirko Bonadei · 5 years ago
  32. 29db239 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  33. e114fb6 Added usrsctp reliablitiy stress test. by Yura Yaroshevich · 5 years ago
  34. 67ac9e8 Prepares RTPSender for extracting RtpSenderEgress by Erik Språng · 5 years ago
  35. 492fdf4 Make rtc_json poisonous in WebRTC by Sam Zackrisson · 5 years ago
  36. 1a61739 Fix MemoryLogWriter so that it always writes the full data. by Björn Terelius · 5 years ago
  37. 53a31f7 Introduce injectable NetEqController interface. by Ivo Creusen · 5 years ago
  38. 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 5 years ago
  39. 49c0880 Revert "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." by Henrik Boström · 5 years ago
  40. 1dac707 Cleanup PacketBuffer tests to use immediate result by Danil Chapovalov · 5 years ago
  41. 4f2783b Speculative Revert: "Use FakeRenderer when fuzzing" by Patrik Höglund · 5 years ago
  42. b394a56 Cleanup of EchoControl interface after downstream fixes by Gustaf Ullberg · 5 years ago
  43. e277bde Add APM test of pre-amplifier gain by Sam Zackrisson · 5 years ago
  44. e914c1e Roll chromium_revision 64883b3ea2..3c5165bebc (709283:709394) by chromium-webrtc-autoroll · 5 years ago
  45. 1dddaa1 [PeerConnection] Use an OperationsChain in PeerConnection for async ops. by Henrik Boström · 5 years ago
  46. ef0e4d0 Roll chromium_revision e1ab9e9b20..64883b3ea2 (709180:709283) by chromium-webrtc-autoroll · 5 years ago
  47. c4af214 Roll chromium_revision d7338c33b2..e1ab9e9b20 (708965:709180) by chromium-webrtc-autoroll · 5 years ago
  48. 3cb6104 AEC3: Support negative delay with external delay estimator by Gustaf Ullberg · 5 years ago
  49. a922904 Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly. by Erik Språng · 5 years ago
  50. b2290f4 Revert "Reset end-of-frame flag in non-VCL packet." by Sergey Silkin · 5 years ago
  51. fc78aac Batches video frame packets when posting to pacer by Erik Språng · 5 years ago
  52. 2040dcf Roll chromium_revision f656c810e4..d7338c33b2 (708845:708965) by chromium-webrtc-autoroll · 5 years ago
  53. eec3919 Remove trial WebRTC-Bwe-ProbeRateFallback by Per Kjellander · 5 years ago
  54. d113ee3 Removes deprecated WebRTC-Bwe-AimdRateControl-NetworkState trial. by Sebastian Jansson · 5 years ago
  55. bd20077 Roll chromium_revision 510c0ca3d7..f656c810e4 (708742:708845) by chromium-webrtc-autoroll · 5 years ago
  56. 0e2b581 RTC_EXPORT webrtc::DesktopCapturerDifferWrapper. by Mirko Bonadei · 5 years ago
  57. c1a8abc Roll chromium_revision b5030084da..510c0ca3d7 (708640:708742) by chromium-webrtc-autoroll · 5 years ago
  58. 4ff1c87 Fix RTC_LOCKABLE RTC_EXPORT order for rtc::Thread. by Mirko Bonadei · 5 years ago
  59. d7bf5c5 Revert "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  60. 6e81567 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 5 years ago
  61. ce1ffcd change PacketBuffer to return it's result rather that use callback by Danil Chapovalov · 5 years ago
  62. 2522b25 Roll chromium_revision 6dc3a51e22..b5030084da (708537:708640) by chromium-webrtc-autoroll · 5 years ago
  63. 6adb0a2 Do not compile webrtc_lib_link_test if is_asan=true. by Mirko Bonadei · 5 years ago
  64. 21bfa40 Update APM config on RuntimeSetting pre amplifier gain change by Sam Zackrisson · 5 years ago
  65. 4f178d0 Fix gtk color-space conversion in peerconnection_client by Niels Möller · 5 years ago
  66. d81a04e Roll chromium_revision c0cca6e419..6dc3a51e22 (708426:708537) by chromium-webrtc-autoroll · 5 years ago
  67. 0ff7c02 Add multipleTouchEnabled for subview of RTCMTLVideoView and RTCEAGLVideoView by CZ Theng · 5 years ago
  68. 27c2936 Implement an OperationsChain, to be used by PeerConnection in follow-up. by Henrik Boström · 5 years ago
  69. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  70. 261fc51 Roll chromium_revision 83bb172f2d..c0cca6e419 (708314:708426) by chromium-webrtc-autoroll · 5 years ago
  71. 8e13e6e Handle no-longer-sticky-in-Q+ WIFI_P2P_CONNECTION_CHANGED_ACTION intent. by Qingsi Wang · 5 years ago
  72. c04792e Component Build support for api/task_queue:task_queue_test. by Mirko Bonadei · 5 years ago
  73. 7058d33 Roll chromium_revision fc69c6f5b4..83bb172f2d (708204:708314) by chromium-webrtc-autoroll · 5 years ago
  74. 05691dd Add possibility to skip check_includes presubmit check. by Mirko Bonadei · 5 years ago
  75. 33678af Roll chromium_revision 9b6351c71d..fc69c6f5b4 (708073:708204) by chromium-webrtc-autoroll · 5 years ago
  76. a043b2b Support case where win32socketserver's window class is not unregistered properly. by Tim Haloun · 5 years ago
  77. 8c51f2e AnalyzeReverseStream with StreamConfig by Gustaf Ullberg · 5 years ago
  78. e76b3ab Add per frame decode time histograms for 4k/HD and VP9/H264 by Johannes Kron · 5 years ago
  79. 13a8e16 Cleanup use of deprecated PacketRouter methods by Erik Språng · 5 years ago
  80. 1272dad Reduce log level of Opus bitrate. by Minyue Li · 5 years ago
  81. d15a028 Hide deprecated SingleThreadedTaskQueueForTest behind an accessor by Danil Chapovalov · 5 years ago
  82. 528a034 Fix fuzzer-found inconsistency in AEC3 config json parsing by Sam Zackrisson · 5 years ago
  83. 5f2fc41 VP9 decoder: replace DCHECK with error message by Ilya Nikolaevskiy · 5 years ago
  84. 0855e2d Delete unused members of MediaReceiverInfo and MediaSenderInfo by Niels Möller · 5 years ago
  85. 85a1000 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue by Danil Chapovalov · 5 years ago
  86. b9014fb Roll chromium_revision b528279c97..9b6351c71d (707828:708073) by chromium-webrtc-autoroll · 5 years ago
  87. ead0ec9 Add firing of OnRemoveTrack and OnRenegotationNeeded during rollback by Eldar Rello · 5 years ago
  88. 4b4713d Roll chromium_revision 8587b26e98..b528279c97 (707701:707828) by chromium-webrtc-autoroll · 5 years ago
  89. eef5e4f Remove dep between test:test_support and rtc_base_approved. by Mirko Bonadei · 5 years ago
  90. c98ff2e Reset end-of-frame flag in non-VCL packet. by Sergey Silkin · 5 years ago
  91. 9cd53b4 Avoid DEPRECATED_SingleThreadedTaskQueueForTesting::CancelTask in VideoAnalyzer by Danil Chapovalov · 5 years ago
  92. e34fb87 Clarify NetworkControl interface: result of each function must be used by Danil Chapovalov · 5 years ago
  93. 9f5ae7b Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting by Danil Chapovalov · 5 years ago
  94. 42b6e2d Change failing rtc::dchecked_cast to rtc::saturated_cast. by Jakob Ivarsson · 5 years ago
  95. 72cc71c Harmonize APM config logging, update config ToString by Sam Zackrisson · 5 years ago
  96. 682dabd Add RTCStatisticsReport.h to WebRTC.framework. by CZ Theng · 5 years ago
  97. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 5 years ago
  98. b9f6902 Store logging streams in a manually linked list instead of std::list by Danil Chapovalov · 5 years ago
  99. 82a3f0a Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask by Danil Chapovalov · 5 years ago
  100. 712b676 Stop using gtest internal macro GTEST_ARRAY_SIZE_ by Danil Chapovalov · 5 years ago