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gerrit-public.fairphone.software / platform / external / webrtc / dbbf41308593afef530c83e73744064eb4ff50a0
  1. dbbf413 Fix use of unitialized value in test by Erik Språng · 4 years, 9 months ago
  2. c66e004 Adding missing RTC_EXPORT for component build. by Mirko Bonadei · 4 years, 9 months ago
  3. 05269ec Rename PacketBuffer tests to follow conventions by Danil Chapovalov · 4 years, 9 months ago
  4. f07003c Avoid Realloc in LibvpxVp8Encoder by Niels Möller · 4 years, 9 months ago
  5. 119e219 AEC3: General cleanup after multichannel changes by Per Åhgren · 4 years, 9 months ago
  6. 2167163 Rewrite the lib link test to just be a binary. by Patrik Höglund · 4 years, 9 months ago
  7. 98872dc AEC3: Update SpectrumBuffer API by Sam Zackrisson · 4 years, 9 months ago
  8. c336dd1 Roll chromium_revision 8a67b116c8..f29003dd01 (707122:707259) by chromium-webrtc-autoroll · 4 years, 9 months ago
  9. 7bf8699 Roll chromium_revision 1d5b803fd3..8a67b116c8 (707010:707122) by chromium-webrtc-autoroll · 4 years, 9 months ago
  10. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 4 years, 9 months ago
  11. e8b962b Roll chromium_revision 60f3e975da..1d5b803fd3 (706891:707010) by chromium-webrtc-autoroll · 4 years, 9 months ago
  12. 88d662a AEC3: Removed some usages of DirectPathFilterDelays by Per Åhgren · 4 years, 9 months ago
  13. ce9da16 Use FakeRenderer when fuzzing by Kuang-che Wu · 4 years, 10 months ago
  14. 785d4c4 AEC3: Add multichannel support in the ERLE estimation by Per Åhgren · 4 years, 9 months ago
  15. db8df17 Add AEC3 config json parsing fuzzer by Sam Zackrisson · 4 years, 9 months ago
  16. 671b403 Split RTPSender into pre- and post-pacer parts. by Erik Språng · 4 years, 9 months ago
  17. eeb79e9 Add a test which breaks if libwebrtc.a don't pull in the right symbols. by Patrik Höglund · 4 years, 9 months ago
  18. 5b74f8d Roll chromium_revision 08af487375..60f3e975da (706400:706891) by chromium-webrtc-autoroll · 4 years, 9 months ago
  19. c71d85b Pass full RtpPacket to RtpVideoStreamReceiver::OnReceivedPayload by Danil Chapovalov · 4 years, 9 months ago
  20. 5b2df17 Width and Height was not associated and provided to decoder for H264 streams which have Nalus before SPS by Shyam Sadhwani · 4 years, 9 months ago
  21. c06aef2 Reland "Use just a lookup map of RTP modules in PacketRouter" by Erik Språng · 4 years, 9 months ago
  22. 5074758 Update DEPS to download the checked-in JDK. by Mirko Bonadei · 4 years, 9 months ago
  23. 0ac52dc Use symbol_level=1 also for MSVC dbg bots. by Mirko Bonadei · 4 years, 9 months ago
  24. fbe84ef Revert "Use just a lookup map of RTP modules in PacketRouter" by Erik Språng · 4 years, 9 months ago
  25. 96f3de0 Use just a lookup map of RTP modules in PacketRouter by Erik Språng · 4 years, 9 months ago
  26. 4970670 Avoid reading outside of memory in WebRtcVad_FindMinimum by Henrik Lundin · 4 years, 9 months ago
  27. dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 4 years, 9 months ago
  28. 5f01bf6 Refactor handling of TransportSequenceNumber in PacketRouter by Erik Språng · 4 years, 9 months ago
  29. a6d7b02 Avoid g_clear_object in pipewire by Tom Anderson · 4 years, 10 months ago
  30. 04671b0 Delete unused method PacedSender::QueueSizePackets by Niels Möller · 4 years, 9 months ago
  31. 7ea9b80 Set StreamDataCountersCallback on construction of RTP modules by Erik Språng · 4 years, 9 months ago
  32. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 10 months ago
  33. 562a37f Increase timeout in test-only helper SendTask back to infinity by Danil Chapovalov · 4 years, 9 months ago
  34. 2bc1ea0 Remove the fileutils hack for good. by Patrik Höglund · 4 years, 9 months ago
  35. 64444bc Roll chromium_revision fbf280c2d2..08af487375 (706299:706400) by chromium-webrtc-autoroll · 4 years, 9 months ago
  36. 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 4 years, 9 months ago
  37. b11c411 Removed unused RTCP methods SendFeedbackPacket and SendNetworkStateEstimate by Per Kjellander · 4 years, 9 months ago
  38. 2f4354e Roll chromium_revision bdc89d87bf..fbf280c2d2 (706176:706299) by chromium-webrtc-autoroll · 4 years, 9 months ago
  39. 2f6e525 Roll chromium_revision 1c1107d4eb..bdc89d87bf (705985:706176) by chromium-webrtc-autoroll · 4 years, 10 months ago
  40. af0aa09 Roll chromium_revision 88a7a88286..1c1107d4eb (705863:705985) by chromium-webrtc-autoroll · 4 years, 10 months ago
  41. 82ed5d1 Replace RtpPacketizerH264::Fragment struct with rtc::ArrayView by Danil Chapovalov · 4 years, 10 months ago
  42. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 4 years, 10 months ago
  43. 82ed2e8 Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender by Sebastian Jansson · 4 years, 10 months ago
  44. 6841d25 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 4 years, 10 months ago
  45. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 4 years, 10 months ago
  46. ffc8452 AEC3: Add support for logging warnings on delay buffer changes by Sam Zackrisson · 4 years, 10 months ago
  47. e8a6bc3 Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"" by Erik Språng · 4 years, 10 months ago
  48. 844600e Put the resources_dir flag into its own target. by Patrik Höglund · 4 years, 10 months ago
  49. c934821 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 4 years, 10 months ago
  50. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 10 months ago
  51. e5d0fe0 Updated VideoStreamEncoder to destroy encoder_queue_ before encoder_switch_experiment_. by philipel · 4 years, 10 months ago
  52. 4ed0b52 Revert "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 4 years, 10 months ago
  53. 8718afb AEC3: Made EchoAudibility multichannel by Per Åhgren · 4 years, 10 months ago
  54. 41478c7 Remove AudioProcessing::gain_control() getter by Sam Zackrisson · 4 years, 10 months ago
  55. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 4 years, 10 months ago
  56. 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 4 years, 10 months ago
  57. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  58. 55c7694 Roll chromium_revision 18d4117247..88a7a88286 (705754:705863) by chromium-webrtc-autoroll · 4 years, 10 months ago
  59. 36d171b Add Ramprakash Jelari to AUTHORS. by Sami Kalliomäki · 4 years, 10 months ago
  60. 17608dc RtpRtcp modules and below: Make media, RTX and FEC SSRCs const by Erik Språng · 4 years, 10 months ago
  61. 2f28370 Move --resources_dir to its right place. by Patrik Höglund · 4 years, 10 months ago
  62. 3f0d8e4 Revert "Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc" by Mirko Bonadei · 4 years, 10 months ago
  63. c122c29 Roll chromium_revision 2e1ac8de05..18d4117247 (705654:705754) by chromium-webrtc-autoroll · 4 years, 10 months ago
  64. d942282 Roll chromium_revision 02833e653c..2e1ac8de05 (705539:705654) by chromium-webrtc-autoroll · 4 years, 10 months ago
  65. f8998cf Add a turn port prune policy to keep the first ready turn port. by Honghai Zhang · 4 years, 10 months ago
  66. ef98ae6 Use GlobalLock to protect logging by Danil Chapovalov · 4 years, 10 months ago
  67. 65c57ff Adds logging of NetworkStateEstimator estimates. by Sebastian Jansson · 4 years, 10 months ago
  68. c6404a1 Add field trial to reduce STUN pings. by Jonas Oreland · 4 years, 10 months ago
  69. b259b0a Roll chromium_revision c1f96a7b93..02833e653c (705365:705539) by chromium-webrtc-autoroll · 4 years, 10 months ago
  70. 24c678f Adds test for loss based controller under cross traffic induced loss. by Sebastian Jansson · 4 years, 10 months ago
  71. 4af7882 Add feature to skip RELAY to non-RELAY connections by Jonas Oreland · 4 years, 10 months ago
  72. 0deef72 Remove deprecated functions in RTPSenderVideo by Danil Chapovalov · 4 years, 10 months ago
  73. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 10 months ago
  74. 41c650b Use bitrate limits provided by encoder. by Sergey Silkin · 4 years, 10 months ago
  75. 5ab79e6 Reland "Implement rollback for setRemoteDescription" by Eldar Rello · 4 years, 10 months ago
  76. 75acef3 Reject invalid spatial index by Kuang-che Wu · 4 years, 10 months ago
  77. d6bb184 Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc by Niels Möller · 4 years, 10 months ago
  78. 8bbf9e2 Roll chromium_revision 002d8b5c6a..c1f96a7b93 (705236:705365) by chromium-webrtc-autoroll · 4 years, 10 months ago
  79. 8be669f AEC3: Add support for multiple channels to the reverb modelling by Per Åhgren · 4 years, 10 months ago
  80. 373b149 Roll chromium_revision da0e48ef9f..002d8b5c6a (705127:705236) by chromium-webrtc-autoroll · 4 years, 10 months ago
  81. 6787f23 Remove AudioProcessing::level_estimator() getter by saza · 4 years, 10 months ago
  82. c67a4d6 Fix WebRTC-Video-MinVideoBitrate for VP9 by Elad Alon · 4 years, 10 months ago
  83. db3d81f Roll chromium_revision 3d7980bda8..da0e48ef9f (705004:705127) by chromium-webrtc-autoroll · 4 years, 10 months ago
  84. d8aff21 Adds support for stopping fake TCP cross traffic. by Sebastian Jansson · 4 years, 10 months ago
  85. 80f53b7 Extend WebRTC-Video-MinVideoBitrate to experiment per-codec by Elad Alon · 4 years, 10 months ago
  86. e62a588 Merging TransportFeedbackAdapter and SendTimeHistory. by Sebastian Jansson · 4 years, 10 months ago
  87. c69c1bb Plot delay feedback in RTCP arrival order. by Björn Terelius · 4 years, 10 months ago
  88. 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 4 years, 10 months ago
  89. 3c918b1 Fix bypass of unnecessary resampling by Gustaf Ullberg · 4 years, 10 months ago
  90. 51bf200 Reduce number of RTPVideoSender::SendVideo parameters by Danil Chapovalov · 4 years, 10 months ago
  91. 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 4 years, 10 months ago
  92. 3b819f3 Move video_sources_.clear() call to CallTest::DestroyStreams by Niels Möller · 4 years, 10 months ago
  93. 7c3b100 Roll chromium_revision 3fcb948181..3d7980bda8 (704895:705004) by chromium-webrtc-autoroll · 4 years, 10 months ago
  94. e6f9bd0 Roll chromium_revision d66030f8c3..3fcb948181 (704779:704895) by chromium-webrtc-autoroll · 4 years, 10 months ago
  95. 3273b5e Roll chromium_revision a1c9c88904..d66030f8c3 (704650:704779) by chromium-webrtc-autoroll · 4 years, 10 months ago
  96. d62ac3f Use fake clock for replay fuzzing by Kuang-che Wu · 4 years, 10 months ago
  97. d0704ce Remove RTCP tests from channel_unittest. by Bjorn A Mellem · 4 years, 10 months ago
  98. ee153c9 Send rtcp::RemoteEstimate and rtcp::TransportFeedback in one packet by Per Kjellander · 4 years, 10 months ago
  99. 9e70f36 Roll chromium_revision 651f5a2987..a1c9c88904 (704530:704650) by chromium-webrtc-autoroll · 4 years, 10 months ago
  100. f17976d Use single thread vp9 decoder for fuzzing by Kuang-che Wu · 4 years, 10 months ago