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gerrit-public.fairphone.software / platform / external / webrtc / a8c2f5180f1d61f99fee6112d1271b179980b302
  1. a8c2f51 Remove unused non-standard RtpEncodingParameters members by Florent Castelli · 5 years ago
  2. 6c0e946 Fix VP8 encoder maxFramerate support by Florent Castelli · 4 years, 11 months ago
  3. 4011de0 Revert "AEC3: Ensure that the high-pass filter effect is on when AEC3 is active" by Per Åhgren · 4 years, 11 months ago
  4. 6a05bb1 AEC3: Add signal dependent mixing before alignment by Per Åhgren · 4 years, 11 months ago
  5. 3a77f93 AEC3: Ensure that the high-pass filter effect is on when AEC3 is active by Per Åhgren · 4 years, 11 months ago
  6. 014dd3c Trials should always be populated in call config. by Erik Språng · 5 years ago
  7. 67d3bc2 Changed parameter name to match the use of it in AEC3 by Per Åhgren · 5 years ago
  8. b5aa0a8 Add IceControllerEvent::ICE_CONTROLLER_RECHECK by Jonas Oreland · 4 years, 11 months ago
  9. 4647412 Make new method pure virtual in the EchoControl interface by Per Åhgren · 5 years ago
  10. 21021f0 NetEq: Fix bug in PLC for multi-channel audio by Henrik Lundin · 4 years, 11 months ago
  11. 5256d8b Refactor FrameGenerator to return VideoFrameBuffer with VideoFrame::UpdateRect by Artem Titov · 4 years, 11 months ago
  12. b2b58d8 AEC3: Adding default AEC3 configurations that are setup specific by Per Åhgren · 4 years, 11 months ago
  13. cf20519 AEC3: Correct the number of render channels in the echo audibility code by Per Åhgren · 4 years, 11 months ago
  14. 00cf34c Refactor DataChannel control out of PeerConnection by Harald Alvestrand · 4 years, 11 months ago
  15. b877e71 Delete FunctorMessageHandler, in tests use alternative ways to post functors by Danil Chapovalov · 4 years, 11 months ago
  16. f2c0818 Minor fixes to ChannelSend. by Mirko Bonadei · 5 years ago
  17. b0db98c Fuzz AEC3 by Sam Zackrisson · 5 years ago
  18. b144c58 Remove deprecated setting for activating multichannel processing by Per Åhgren · 5 years ago
  19. efa3f76 Moves SampleStats and EventRateCounter to rtc_base/numerics by Sebastian Jansson · 4 years, 11 months ago
  20. 865a74e Roll chromium_revision a9c1e4afb9..3f97848513 (720171:720272) by chromium-webrtc-autoroll · 4 years, 11 months ago
  21. d003662 Move SendBindingResponse to Connection by Jonas Oreland · 4 years, 11 months ago
  22. 8931345 Take FunctionView rather than any functor reference in the rtc::Thread::Invoke by Danil Chapovalov · 4 years, 11 months ago
  23. 39cf3c7 Clean up the NetEqFactory API. by Ivo Creusen · 5 years ago
  24. 2d02c94 NetEQ: fix initial decoder frame length. by Alessio Bazzica · 4 years, 11 months ago
  25. 331b00b Roll chromium_revision 26d0995106..a9c1e4afb9 (720053:720171) by chromium-webrtc-autoroll · 4 years, 11 months ago
  26. 499b3b6 In RtpDepacketizerAV1 use aggregation header to detect key frames by Danil Chapovalov · 5 years ago
  27. c7a3b08 Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_. by Mirko Bonadei · 5 years ago
  28. a3cd717 Remove WebRTC-Bwe-CongestionWindowDownlinkDelay. by Mirko Bonadei · 5 years ago
  29. 62c3936 Roll chromium_revision 8955e0d38a..26d0995106 (719771:720053) by chromium-webrtc-autoroll · 5 years ago
  30. fe7ce1c Fix ErrorProne MultiVariableDeclaration. by Mirko Bonadei · 5 years ago
  31. 0682ca9 Use AV1 packetizer/depacketizer for AV1 bitstreams by Danil Chapovalov · 5 years ago
  32. 9dc209a Add ability to disable detailed error message in RTC_CHECKs by Artem Titov · 5 years ago
  33. 9750e84 AEC3:Turning off default downmix in surround alignment by Per Åhgren · 5 years ago
  34. 253d50f Add new Stun utility functions by Jonas Oreland · 5 years ago
  35. 2dec496 Add directive to make TRACE_EVENT macros optional. by Doudou Kisabaka · 5 years ago
  36. b8306cc Remove temporary 8-bit H264 HDR fix by Johannes Kron · 5 years ago
  37. 096a46f Implement AV1 RtpPacketizer class by Danil Chapovalov · 5 years ago
  38. 4314a49 Implements a task-queue based PacedSender, wires it up for field trials by Erik Språng · 5 years ago
  39. 5314b13 Fix undefined-shift in RtpDepacketizerAv1::AssembleFrame by Danil Chapovalov · 5 years ago
  40. bfcb6c3 Add rtt estimate EventBasedExponentialMovingAverage to Connection by Jonas Oreland · 5 years ago
  41. 6532fc6 Roll chromium_revision 2b82941a12..8955e0d38a (719656:719771) by chromium-webrtc-autoroll · 5 years ago
  42. e1611a0 Replace template_util.h with C++14 STL methods by Steve Anton · 5 years ago
  43. b3fb339 Roll chromium_revision 65c32b57c2..2b82941a12 (719106:719656) by chromium-webrtc-autoroll · 5 years ago
  44. 3f75209 Revert "Remove temporary workaround for generate_licenses." by Yves Gerey · 5 years ago
  45. d08bb1e Propagate absolute capture time through video receiving side. by Ruslan Burakov · 5 years ago
  46. 7968530 Removes caching SimulcastEncoderAdapter::GetEncoderInfo() by Erik Språng · 5 years ago
  47. 840394c Fix bw_limited_resolution in SendStatisticsProxy GetStats by Ilya Nikolaevskiy · 5 years ago
  48. b529b7a Add string<->VideoCodecType conversion for all codec types. by Danil Chapovalov · 5 years ago
  49. 5cef9c3 Revert "Add support for RtpEncodingParameters::max_framerate" by Florent Castelli · 5 years ago
  50. 8861d02 Restore tests that were accidently deleted during refactoring by Artem Titov · 5 years ago
  51. 9f9e20a Fix errorprone issues preventing Chromium Roll. by Mirko Bonadei · 5 years ago
  52. 6565681 Turn off Goma for Linux GCC. by Patrik Höglund · 5 years ago
  53. 26cc5e6 Corrected the aggregation of AGC choices and add fallback solution by Per Åhgren · 5 years ago
  54. 17e4c58 Adding parametrization of the AEC3 howling mitigation behavior by Per Åhgren · 5 years ago
  55. 98e745b make Connection::port() protected by Jonas Oreland · 5 years ago
  56. b1ccae2 Reland "Fixes dynamic mode pacing issues." by Erik Språng · 5 years ago
  57. dc36829 Add VideoCodecType::kVideoCodecAV1 value by Danil Chapovalov · 5 years ago
  58. e14cb99 Correct/update the activation of the multi-channel processing in APM by Per Åhgren · 5 years ago
  59. c363982 Convert proxy.h helper classes to variadic templates by Steve Anton · 5 years ago
  60. 2a6b3b1 Correcting the analog AGC re-initialization at device changes by Per Åhgren · 5 years ago
  61. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
  62. 2b4bd97 Fix fuzzer-found bug in fuzzer by Sam Zackrisson · 5 years ago
  63. 44d7ec0 Add Opus-only audio codec factories by Karl Wiberg · 5 years ago
  64. 3967389 Revert "Fixes dynamic mode pacing issues." by Erik Språng · 5 years ago
  65. 6a4a146 Add ability to strip out logging messages from the binary by Artem Titov · 5 years ago
  66. 68c6572 Add a CreateNetEq method that takes an AudioDecoderFactory by Ivo Creusen · 5 years ago
  67. fdaba6c Use std::atomic for RefCounter by Danil Chapovalov · 5 years ago
  68. ef4ab76 Roll chromium_revision bdce346064..65c32b57c2 (718772:719106) by chromium-webrtc-autoroll · 5 years ago
  69. a88655d NetEQ RTP play: textlog to stderr as option by Alessio Bazzica · 5 years ago
  70. 27bd76b DCHECKing for deprecated 8kHz support in AGC and changing fuzzer by Per Åhgren · 5 years ago
  71. a101a4f Reland "Add IvfVideoFrameGenerator" by Artem Titov · 5 years ago
  72. d5e2f21 VideoRtpTrackSource: implement encoded source methods. by Markus Handell · 5 years ago
  73. 72e6cb0 Fixes dynamic mode pacing issues. by Erik Språng · 5 years ago
  74. 13ea34f Revert "Add IvfVideoFrameGenerator" by Steve Anton · 5 years ago
  75. 353a718 Address failing wpt test cases for the rollback feature by Eldar Rello · 5 years ago
  76. 6620506 Roll chromium_revision e0dd604cfd..bdce346064 (718623:718772) by chromium-webrtc-autoroll · 5 years ago
  77. 15be528 Add support for RtpEncodingParameters::max_framerate by Florent Castelli · 5 years ago
  78. f534a64 AEC3: Sub-band nearend detector by Gustaf Ullberg · 5 years ago
  79. 7f44505 Allow min start bitrate to be lower than min bitrate. by Sergey Silkin · 5 years ago
  80. eee1a49 Roll chromium_revision 0ec8ef3c9f..e0dd604cfd (718517:718623) by chromium-webrtc-autoroll · 5 years ago
  81. c7a46c4 Fix VideoStreamEncoder to not reference encoded data from the RunPostEncode task. by philipel · 5 years ago
  82. 712a26f Add IvfVideoFrameGenerator by Artem Titov · 5 years ago
  83. 63dced9 Add class for ExponentialMovingAverage by Jonas Oreland · 5 years ago
  84. fba4481 Make it possible to inject a custom NetEqFactory from the java interface. by Ivo Creusen · 5 years ago
  85. c421f3e Makes sprang@ owner in modules/pacing by Erik Språng · 5 years ago
  86. 80b2806 Fixing a buffer overflow in Merge::Downsample by Henrik Lundin · 5 years ago
  87. 00cc836 Makes all of RtpVideoSenderTest use simulated time by Erik Språng · 5 years ago
  88. 912b3b8 Make rtc::Thread a TaskQueue by Danil Chapovalov · 5 years ago
  89. 2aaf4af Semiautomatic attempt to clean test/BUILD.gn deps by Artem Titov · 5 years ago
  90. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  91. 429d8fe Add fuzzer test for RtpDepacketizerAv1::AssembleFrame function by Danil Chapovalov · 5 years ago
  92. eac63e7 Remove temporary workaround for generate_licenses. by Yves Gerey · 5 years ago
  93. 31b01c0 Fuzz APM float interface with up to 8 channels by Sam Zackrisson · 5 years ago
  94. 2ad66ec Roll chromium_revision 126f20ede5..0ec8ef3c9f (718412:718517) by chromium-webrtc-autoroll · 5 years ago
  95. 51868f5 Revert "Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."" by Mirko Bonadei · 5 years ago
  96. 8994c8b Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium." by Mirko Bonadei · 5 years ago
  97. e85faf9 Roll chromium_revision 53149b644c..126f20ede5 (718304:718412) by chromium-webrtc-autoroll · 5 years ago
  98. 77dc199 Changed the digital AGC1 gain to properly support multichannel by Per Åhgren · 5 years ago
  99. 3af0cd8 Revert "Make webrtc_fuzzer_main depend on webrtc_component in Chromium." by Mirko Bonadei · 5 years ago
  100. 1833a0c Roll chromium_revision 5395db8bba..53149b644c (718200:718304) by chromium-webrtc-autoroll · 5 years ago