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gerrit-public.fairphone.software / platform / external / webrtc / 1272dade560b7fa707acbfb9bd30a98cc5bba0b4
  1. f8998cf Add a turn port prune policy to keep the first ready turn port. by Honghai Zhang · 5 years ago
  2. ef98ae6 Use GlobalLock to protect logging by Danil Chapovalov · 5 years ago
  3. 65c57ff Adds logging of NetworkStateEstimator estimates. by Sebastian Jansson · 5 years ago
  4. c6404a1 Add field trial to reduce STUN pings. by Jonas Oreland · 5 years ago
  5. b259b0a Roll chromium_revision c1f96a7b93..02833e653c (705365:705539) by chromium-webrtc-autoroll · 5 years ago
  6. 24c678f Adds test for loss based controller under cross traffic induced loss. by Sebastian Jansson · 5 years ago
  7. 4af7882 Add feature to skip RELAY to non-RELAY connections by Jonas Oreland · 5 years ago
  8. 0deef72 Remove deprecated functions in RTPSenderVideo by Danil Chapovalov · 5 years ago
  9. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  10. 41c650b Use bitrate limits provided by encoder. by Sergey Silkin · 5 years ago
  11. 5ab79e6 Reland "Implement rollback for setRemoteDescription" by Eldar Rello · 5 years ago
  12. 75acef3 Reject invalid spatial index by Kuang-che Wu · 5 years ago
  13. d6bb184 Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc by Niels Möller · 5 years ago
  14. 8bbf9e2 Roll chromium_revision 002d8b5c6a..c1f96a7b93 (705236:705365) by chromium-webrtc-autoroll · 5 years ago
  15. 8be669f AEC3: Add support for multiple channels to the reverb modelling by Per Åhgren · 5 years ago
  16. 373b149 Roll chromium_revision da0e48ef9f..002d8b5c6a (705127:705236) by chromium-webrtc-autoroll · 5 years ago
  17. 6787f23 Remove AudioProcessing::level_estimator() getter by saza · 5 years ago
  18. c67a4d6 Fix WebRTC-Video-MinVideoBitrate for VP9 by Elad Alon · 5 years ago
  19. db3d81f Roll chromium_revision 3d7980bda8..da0e48ef9f (705004:705127) by chromium-webrtc-autoroll · 5 years ago
  20. d8aff21 Adds support for stopping fake TCP cross traffic. by Sebastian Jansson · 5 years ago
  21. 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 5 years ago
  22. e62a588 Merging TransportFeedbackAdapter and SendTimeHistory. by Sebastian Jansson · 5 years ago
  23. c69c1bb Plot delay feedback in RTCP arrival order. by Björn Terelius · 5 years ago
  24. 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 5 years ago
  25. 3c918b1 Fix bypass of unnecessary resampling by Gustaf Ullberg · 5 years ago
  26. 51bf200 Reduce number of RTPVideoSender::SendVideo parameters by Danil Chapovalov · 5 years ago
  27. 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 5 years ago
  28. 3b819f3 Move video_sources_.clear() call to CallTest::DestroyStreams by Niels Möller · 5 years ago
  29. 7c3b100 Roll chromium_revision 3fcb948181..3d7980bda8 (704895:705004) by chromium-webrtc-autoroll · 5 years ago
  30. e6f9bd0 Roll chromium_revision d66030f8c3..3fcb948181 (704779:704895) by chromium-webrtc-autoroll · 5 years ago
  31. 3273b5e Roll chromium_revision a1c9c88904..d66030f8c3 (704650:704779) by chromium-webrtc-autoroll · 5 years ago
  32. d62ac3f Use fake clock for replay fuzzing by Kuang-che Wu · 5 years ago
  33. d0704ce Remove RTCP tests from channel_unittest. by Bjorn A Mellem · 5 years ago
  34. ee153c9 Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet by Per Kjellander · 5 years ago
  35. 9e70f36 Roll chromium_revision 651f5a2987..a1c9c88904 (704530:704650) by chromium-webrtc-autoroll · 5 years ago
  36. f17976d Use single thread vp9 decoder for fuzzing by Kuang-che Wu · 5 years ago
  37. 45eb135 Remove the unused `receive_timestamp` arg to NetEq::InsertPacket by Karl Wiberg · 5 years ago
  38. c466f08 Cap vp9 fuzzer frame size to prevent OOM by Kuang-che Wu · 5 years ago
  39. cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
  40. 9afdddf Enable capturing from camera in PC framework by Artem Titov · 5 years ago
  41. 1699981 Add void::RtcpFeedbackSenderInterface::SendCombinedRtcpPacket by Per Kjellander · 5 years ago
  42. 03f4b36 Roll chromium_revision d9b4f45e42..651f5a2987 (704251:704530) by chromium-webrtc-autoroll · 5 years ago
  43. cbbfd08 Replace virtual RtcpPacket::SetSenderSsrc with base member by Danil Chapovalov · 5 years ago
  44. 907f154 Revert "Implement rollback for setRemoteDescription" by Alex Loiko · 5 years ago
  45. 28214cd Fix handling of large packets in RtxReceiveStream by Niels Möller · 5 years ago
  46. 8675eee Bypass unnecessary resampling. by Gustaf Ullberg · 5 years ago
  47. ba700de Add missing dependencies to the static library. by Mirko Bonadei · 5 years ago
  48. 066c2ab Roll chromium_revision 8e1616e4fc..d9b4f45e42 (704145:704251) by chromium-webrtc-autoroll · 5 years ago
  49. 16d4c4d Implement rollback for setRemoteDescription by Eldar Rello · 5 years ago
  50. 5963c7c Count disabled due to low bw streams or layers as bw limited quality in GetStats by Ilya Nikolaevskiy · 5 years ago
  51. 955f8fd Add virtual method rtcp::RtcpPacket::SetSenderSsrc by Per Kjellander · 5 years ago
  52. 6f41f8e Roll chromium_revision b2d00427a6..8e1616e4fc (703937:704145) by chromium-webrtc-autoroll · 5 years ago
  53. f3f03e2 Removing outdated tests. by Alex Loiko · 5 years ago
  54. f980725 AEC3: Send the spectral power estimates for all channels to AecState by Per Åhgren · 5 years ago
  55. d9755ee Delete large up-front allocation in LibvpxVp8Encoder::InitEncode by Niels Möller · 5 years ago
  56. 422b9e0 Run fullband processing at output rate on ARM by Gustaf Ullberg · 5 years ago
  57. 1d3008b AEC3: Remove redundant class by Per Åhgren · 5 years ago
  58. 9ddd729 Add Duration field to EventRateCounter by Evan Shrubsole · 5 years ago
  59. 0169a3e Delete AecState::EchoPathGain() by Sam Zackrisson · 5 years ago
  60. e1092c0 Roll chromium_revision a78cc9b4cc..b2d00427a6 (703818:703937) by chromium-webrtc-autoroll · 5 years ago
  61. 6e9395c Roll chromium_revision baa7b58596..a78cc9b4cc (703669:703818) by chromium-webrtc-autoroll · 5 years ago
  62. f77b939 Makes render time > decode time in VideoFrameMatcher. by Sebastian Jansson · 5 years ago
  63. 46b0140 Update filter analyzer for multi channel by Sam Zackrisson · 5 years ago
  64. 43bd760 Fix build errors of RTCAudioDeviceTests by Byoungchan Lee · 5 years ago
  65. cfe5e2a Stop using goma for MSVC bots. by Mirko Bonadei · 5 years ago
  66. fa77ba6 SetStreams API of RtpSender wrapped for iOS and Android by Cyril Lashkevich · 5 years ago
  67. 999afa9 Fix cropping in H264 decoder wrapper. by Sergey Silkin · 5 years ago
  68. 7f9a0f3 Roll chromium_revision 977e732442..baa7b58596 (703537:703669) by chromium-webrtc-autoroll · 5 years ago
  69. d46d1e9 Add #COMPONENT to WebRTC. by Patrik Höglund · 5 years ago
  70. e93b1fe Improve bitstream dumping logic to handle multiple SLs correctly by Ilya Nikolaevskiy · 5 years ago
  71. b4161d3 AEC3: Add multichannel support to the residual echo estimator by Per Åhgren · 5 years ago
  72. 7e6abf0 Roll chromium_revision 5ac2340a23..977e732442 (703358:703537) by chromium-webrtc-autoroll · 5 years ago
  73. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
  74. a639f7a Roll chromium_revision 10156469d6..5ac2340a23 (703248:703358) by chromium-webrtc-autoroll · 5 years ago
  75. 7c06777 Cleanup includes in modules/include/module_common_types.h by Danil Chapovalov · 5 years ago
  76. 0824c6f Delete voice_detection() pointer to submodule by Sam Zackrisson · 5 years ago
  77. 24d251f Add 100 ms network delay to the SupportsFlexFEC* tests. by Björn Terelius · 5 years ago
  78. 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
  79. 99a2096 Added support for skipping get_audio events, adding dummy packets and setting a field trial string. by Ivo Creusen · 5 years ago
  80. 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
  81. db0b3bc Roll chromium_revision 35431c5114..10156469d6 (703133:703248) by chromium-webrtc-autoroll · 5 years ago
  82. b441acf AEC3: Add support in the echo subtractor for handling multiple channels by Per Åhgren · 5 years ago
  83. d21db5d Roll chromium_revision e2b55cc552..35431c5114 (703005:703133) by chromium-webrtc-autoroll · 5 years ago
  84. 0e0a04c Roll chromium_revision b5ead1daa2..e2b55cc552 (702047:703005) by chromium-webrtc-autoroll · 5 years ago
  85. 2b84dad Fixed issue with H264 packet buffer where it was not detecting presence of sps/pps for idr frames by Shyam Sadhwani · 5 years ago
  86. 4f2e940 ACM: Adding support for more than 2 channels in the send pipeline by Per Åhgren · 5 years ago
  87. dc34a25 Adds RTPSenderVideo::Config struct with red/ulpfec config by Erik Språng · 5 years ago
  88. b9bfe65 Delete VCMEncodedFrame::VerifyAndAllocate by Niels Möller · 5 years ago
  89. 7536bc5 Account for IP and UDP headers in emulated network by Niels Möller · 5 years ago
  90. ed8eadc Update RTC_LOGs in DtlsTransport to be able to distinguish errors. by Henrik Boström · 5 years ago
  91. f83d0ef Revert "Remove an old hack from test_main_lib.cc." by Patrik Höglund · 5 years ago
  92. 82a5100 Replacing /target:target with /target in BUILD autofix. by Sebastian Jansson · 5 years ago
  93. ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 5 years ago
  94. 79f3287 Cleanup of simple TODO(srte) comments. by Sebastian Jansson · 5 years ago
  95. 5114a92 Remove an old hack from test_main_lib.cc. by Patrik Höglund · 5 years ago
  96. 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
  97. 78c82a4 Adds trial to always start probes with a small padding packet. by Erik Språng · 5 years ago
  98. 608083b Reset QP sum when QP is not reported on decoded frame. by Mirta Dvornicic · 5 years ago
  99. 6cf554e Reduces locking in RtpSenderVideo. by Erik Språng · 5 years ago
  100. f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago