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gerrit-public.fairphone.software / platform / external / webrtc / 4b4713db32a3ccec3984df77b4317082d6f42b70
  1. 4b4713d Roll chromium_revision 8587b26e98..b528279c97 (707701:707828) by chromium-webrtc-autoroll · 4 years, 9 months ago
  2. eef5e4f Remove dep between test:test_support and rtc_base_approved. by Mirko Bonadei · 4 years, 9 months ago
  3. c98ff2e Reset end-of-frame flag in non-VCL packet. by Sergey Silkin · 4 years, 9 months ago
  4. 9cd53b4 Avoid DEPRECATED_SingleThreadedTaskQueueForTesting::CancelTask in VideoAnalyzer by Danil Chapovalov · 4 years, 9 months ago
  5. e34fb87 Clarify NetworkControl interface: result of each function must be used by Danil Chapovalov · 4 years, 9 months ago
  6. 9f5ae7b Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting by Danil Chapovalov · 4 years, 9 months ago
  7. 42b6e2d Change failing rtc::dchecked_cast to rtc::saturated_cast. by Jakob Ivarsson · 4 years, 9 months ago
  8. 72cc71c Harmonize APM config logging, update config ToString by Sam Zackrisson · 4 years, 9 months ago
  9. 682dabd Add RTCStatisticsReport.h to WebRTC.framework. by CZ Theng · 4 years, 9 months ago
  10. 03fbace Remove apm_helpers, consolidate audio config in WebRtcVoiceEngine by Sam Zackrisson · 4 years, 9 months ago
  11. b9f6902 Store logging streams in a manually linked list instead of std::list by Danil Chapovalov · 4 years, 9 months ago
  12. 82a3f0a Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask by Danil Chapovalov · 4 years, 9 months ago
  13. 712b676 Stop using gtest internal macro GTEST_ARRAY_SIZE_ by Danil Chapovalov · 4 years, 9 months ago
  14. 2b739f0 Roll chromium_revision db43fd3604..8587b26e98 (707597:707701) by chromium-webrtc-autoroll · 4 years, 9 months ago
  15. 3663ed3 Move rtc_base/ignore_wundef.h to its own target. by Mirko Bonadei · 4 years, 9 months ago
  16. d441e50 Roll chromium_revision 8b1da07aa3..db43fd3604 (707493:707597) by chromium-webrtc-autoroll · 4 years, 9 months ago
  17. fdf54f2 Stop pairing local relay candidates with remote mDNS candidates. by Qingsi Wang · 4 years, 9 months ago
  18. 41595dd Roll chromium_revision 0d7020346c..8b1da07aa3 (707387:707493) by chromium-webrtc-autoroll · 4 years, 9 months ago
  19. 0e3485c Revert "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true." by Mirko Bonadei · 4 years, 9 months ago
  20. db94d14 Roll chromium_revision f29003dd01..0d7020346c (707259:707387) by chromium-webrtc-autoroll · 4 years, 9 months ago
  21. 6e5433c AEC3: Multi channel ERL estimator by Sam Zackrisson · 4 years, 9 months ago
  22. 33ed882 Update the minimum bitrate when a stream allocation is removed. by Jakob Ivarsson · 4 years, 9 months ago
  23. 03bc15c Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true. by Mirko Bonadei · 4 years, 10 months ago
  24. 1d60052 Delete ApmPublicSubmodules, fix ApmPrivateSubmodules, ApmSubmoduleStates by saza · 4 years, 9 months ago
  25. 3f7e0ed Add option to make first scale factor depend on input resolution. by Åsa Persson · 4 years, 9 months ago
  26. 89e130a Reland "Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc" by Niels Möller · 4 years, 10 months ago
  27. 55d19e5 Add gustaf to audio/OWNERS by Gustaf Ullberg · 4 years, 9 months ago
  28. 4aae11d Remove redundant fields in PacketBuffer by Danil Chapovalov · 4 years, 9 months ago
  29. dbbf413 Fix use of unitialized value in test by Erik Språng · 4 years, 9 months ago
  30. c66e004 Adding missing RTC_EXPORT for component build. by Mirko Bonadei · 4 years, 9 months ago
  31. 05269ec Rename PacketBuffer tests to follow conventions by Danil Chapovalov · 4 years, 9 months ago
  32. f07003c Avoid Realloc in LibvpxVp8Encoder by Niels Möller · 4 years, 9 months ago
  33. 119e219 AEC3: General cleanup after multichannel changes by Per Åhgren · 4 years, 9 months ago
  34. 2167163 Rewrite the lib link test to just be a binary. by Patrik Höglund · 4 years, 9 months ago
  35. 98872dc AEC3: Update SpectrumBuffer API by Sam Zackrisson · 4 years, 9 months ago
  36. c336dd1 Roll chromium_revision 8a67b116c8..f29003dd01 (707122:707259) by chromium-webrtc-autoroll · 4 years, 9 months ago
  37. 7bf8699 Roll chromium_revision 1d5b803fd3..8a67b116c8 (707010:707122) by chromium-webrtc-autoroll · 4 years, 9 months ago
  38. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 4 years, 9 months ago
  39. e8b962b Roll chromium_revision 60f3e975da..1d5b803fd3 (706891:707010) by chromium-webrtc-autoroll · 4 years, 9 months ago
  40. 88d662a AEC3: Removed some usages of DirectPathFilterDelays by Per Åhgren · 4 years, 9 months ago
  41. ce9da16 Use FakeRenderer when fuzzing by Kuang-che Wu · 4 years, 10 months ago
  42. 785d4c4 AEC3: Add multichannel support in the ERLE estimation by Per Åhgren · 4 years, 9 months ago
  43. db8df17 Add AEC3 config json parsing fuzzer by Sam Zackrisson · 4 years, 9 months ago
  44. 671b403 Split RTPSender into pre- and post-pacer parts. by Erik Språng · 4 years, 9 months ago
  45. eeb79e9 Add a test which breaks if libwebrtc.a don't pull in the right symbols. by Patrik Höglund · 4 years, 9 months ago
  46. 5b74f8d Roll chromium_revision 08af487375..60f3e975da (706400:706891) by chromium-webrtc-autoroll · 4 years, 9 months ago
  47. c71d85b Pass full RtpPacket to RtpVideoStreamReceiver::OnReceivedPayload by Danil Chapovalov · 4 years, 9 months ago
  48. 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
  49. c06aef2 Reland "Use just a lookup map of RTP modules in PacketRouter" by Erik Språng · 4 years, 9 months ago
  50. 5074758 Update DEPS to download the checked-in JDK. by Mirko Bonadei · 4 years, 9 months ago
  51. 0ac52dc Use symbol_level=1 also for MSVC dbg bots. by Mirko Bonadei · 4 years, 9 months ago
  52. fbe84ef Revert "Use just a lookup map of RTP modules in PacketRouter" by Erik Språng · 4 years, 9 months ago
  53. 96f3de0 Use just a lookup map of RTP modules in PacketRouter by Erik Språng · 4 years, 9 months ago
  54. 4970670 Avoid reading outside of memory in WebRtcVad_FindMinimum by Henrik Lundin · 4 years, 9 months ago
  55. dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 4 years, 9 months ago
  56. 5f01bf6 Refactor handling of TransportSequenceNumber in PacketRouter by Erik Språng · 4 years, 9 months ago
  57. a6d7b02 Avoid g_clear_object in pipewire by Tom Anderson · 4 years, 10 months ago
  58. 04671b0 Delete unused method PacedSender::QueueSizePackets by Niels Möller · 4 years, 9 months ago
  59. 7ea9b80 Set StreamDataCountersCallback on construction of RTP modules by Erik Språng · 4 years, 9 months ago
  60. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 4 years, 10 months ago
  61. 562a37f Increase timeout in test-only helper SendTask back to infinity by Danil Chapovalov · 4 years, 9 months ago
  62. 2bc1ea0 Remove the fileutils hack for good. by Patrik Höglund · 4 years, 9 months ago
  63. 64444bc Roll chromium_revision fbf280c2d2..08af487375 (706299:706400) by chromium-webrtc-autoroll · 4 years, 9 months ago
  64. 0bad15f Remove the noise_suppression() pointer to submodule interface by saza · 4 years, 9 months ago
  65. b11c411 Removed unused RTCP methods SendFeedbackPacket and SendNetworkStateEstimate by Per Kjellander · 4 years, 9 months ago
  66. 2f4354e Roll chromium_revision bdc89d87bf..fbf280c2d2 (706176:706299) by chromium-webrtc-autoroll · 4 years, 9 months ago
  67. 2f6e525 Roll chromium_revision 1c1107d4eb..bdc89d87bf (705985:706176) by chromium-webrtc-autoroll · 4 years, 9 months ago
  68. af0aa09 Roll chromium_revision 88a7a88286..1c1107d4eb (705863:705985) by chromium-webrtc-autoroll · 4 years, 10 months ago
  69. 82ed5d1 Replace RtpPacketizerH264::Fragment struct with rtc::ArrayView by Danil Chapovalov · 4 years, 10 months ago
  70. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 4 years, 10 months ago
  71. 82ed2e8 Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender by Sebastian Jansson · 4 years, 10 months ago
  72. 6841d25 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 4 years, 10 months ago
  73. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 4 years, 10 months ago
  74. ffc8452 AEC3: Add support for logging warnings on delay buffer changes by Sam Zackrisson · 4 years, 10 months ago
  75. e8a6bc3 Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"" by Erik Språng · 4 years, 10 months ago
  76. 844600e Put the resources_dir flag into its own target. by Patrik Höglund · 4 years, 10 months ago
  77. c934821 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 4 years, 10 months ago
  78. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 10 months ago
  79. e5d0fe0 Updated VideoStreamEncoder to destroy encoder_queue_ before encoder_switch_experiment_. by philipel · 4 years, 10 months ago
  80. 4ed0b52 Revert "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 4 years, 10 months ago
  81. 8718afb AEC3: Made EchoAudibility multichannel by Per Åhgren · 4 years, 10 months ago
  82. 41478c7 Remove AudioProcessing::gain_control() getter by Sam Zackrisson · 4 years, 10 months ago
  83. eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 4 years, 10 months ago
  84. 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 4 years, 10 months ago
  85. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  86. 55c7694 Roll chromium_revision 18d4117247..88a7a88286 (705754:705863) by chromium-webrtc-autoroll · 4 years, 10 months ago
  87. 36d171b Add Ramprakash Jelari to AUTHORS. by Sami Kalliomäki · 4 years, 10 months ago
  88. 17608dc RtpRtcp modules and below: Make media, RTX and FEC SSRCs const by Erik Språng · 4 years, 10 months ago
  89. 2f28370 Move --resources_dir to its right place. by Patrik Höglund · 4 years, 10 months ago
  90. 3f0d8e4 Revert "Delete methods EncodedImage::Allocate and EncodedImageBufferInterface::Realloc" by Mirko Bonadei · 4 years, 10 months ago
  91. c122c29 Roll chromium_revision 2e1ac8de05..18d4117247 (705654:705754) by chromium-webrtc-autoroll · 4 years, 10 months ago
  92. d942282 Roll chromium_revision 02833e653c..2e1ac8de05 (705539:705654) by chromium-webrtc-autoroll · 4 years, 10 months ago
  93. f8998cf Add a turn port prune policy to keep the first ready turn port. by Honghai Zhang · 4 years, 10 months ago
  94. ef98ae6 Use GlobalLock to protect logging by Danil Chapovalov · 4 years, 10 months ago
  95. 65c57ff Adds logging of NetworkStateEstimator estimates. by Sebastian Jansson · 4 years, 10 months ago
  96. c6404a1 Add field trial to reduce STUN pings. by Jonas Oreland · 4 years, 10 months ago
  97. b259b0a Roll chromium_revision c1f96a7b93..02833e653c (705365:705539) by chromium-webrtc-autoroll · 4 years, 10 months ago
  98. 24c678f Adds test for loss based controller under cross traffic induced loss. by Sebastian Jansson · 4 years, 10 months ago
  99. 4af7882 Add feature to skip RELAY to non-RELAY connections by Jonas Oreland · 4 years, 10 months ago
  100. 0deef72 Remove deprecated functions in RTPSenderVideo by Danil Chapovalov · 4 years, 10 months ago