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gerrit-public.fairphone.software / platform / external / webrtc / ace2a2bcce455c56dfcf12c1b788473f172245ae
  1. 368d002 Roll chromium_revision dbd1569418..31d9542abc (697157:697288) by chromium-webrtc-autoroll · 5 years ago
  2. 9fa8ef1 absl::make_unique presubmit check. by Mirko Bonadei · 5 years ago
  3. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  4. 809198e Fix minor regression caused by a8336d3 by Evan Shrubsole · 5 years ago
  5. 7d00342 Remove old packet socket factory header. by Patrik Höglund · 5 years ago
  6. e1b7777 Removing deprecated min_pacing_rate alias in StreamsConfig. by Sebastian Jansson · 5 years ago
  7. 4a822f4 Roll chromium_revision 2e4ccff8a8..dbd1569418 (696956:697157) by chromium-webrtc-autoroll · 5 years ago
  8. 2c6ea52 In TaskQueueTest::PostDelayedAfterDesctruct increase timeout by Danil Chapovalov · 5 years ago
  9. c1c6284 New (empty) build target api:media_stream_interface by Niels Möller · 5 years ago
  10. 1722182 Roll chromium_revision 3cf04dec00..2e4ccff8a8 (696812:696956) by chromium-webrtc-autoroll · 5 years ago
  11. 7262fc2 Refactor Rtp Receivers to accept SSRC 0. by Saurav Das · 5 years ago
  12. 3d16474 in RtcpTransciever use lambdas with move capture. by Danil Chapovalov · 5 years ago
  13. 3462793 Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812) by chromium-webrtc-autoroll · 5 years ago
  14. 68ef259 Delete deprecated rtc_event.h file by Danil Chapovalov · 5 years ago
  15. f5dec1c Implement Dependency Descriptor reader by Danil Chapovalov · 5 years ago
  16. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 5 years ago
  17. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 5 years ago
  18. 1b6a30d Update WebRTC's C++ style guide to reflect the switch to C++14. by Mirko Bonadei · 5 years ago
  19. a740142 Refactor LossNotificationController to not use VCMPacket by Niels Möller · 5 years ago
  20. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
  21. c4e80ad Delete forward declarations from peer_connection_interface.h by Niels Möller · 5 years ago
  22. 7af1bb3 Roll chromium_revision 9f15168729..1d12ff693d (696593:696696) by chromium-webrtc-autoroll · 5 years ago
  23. fcbe407 Adding more refined control over choice of band-splitting by Per Åhgren · 5 years ago
  24. ec06ebd Roll chromium_revision 9004bcf36a..9f15168729 (696490:696593) by chromium-webrtc-autoroll · 5 years ago
  25. 0dd37ce Roll chromium_revision 4740202690..9004bcf36a (696373:696490) by chromium-webrtc-autoroll · 5 years ago
  26. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  27. 6a6eb61 Roll chromium_revision f7cd88eb51..4740202690 (696270:696373) by chromium-webrtc-autoroll · 5 years ago
  28. e78fd80 New class DummyPeerConnection by Niels Möller · 5 years ago
  29. 3873927 Fix time units in plotted charts by Artem Titov · 5 years ago
  30. 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 5 years ago
  31. 0a7d5d8 Set console window NOTOPMOST flag after WindowFinderTest.FindDrawerWindow on Windows by Kimmo Kinnunen · 5 years ago
  32. 01be33b Using lambdas instead of rtc::Bind in BaseChannel. by Sebastian Jansson · 5 years ago
  33. 262bbae Fix rare audioLevel flake in RTCStatsIntegrationTest. by Henrik Boström · 5 years ago
  34. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  35. 5f15f86 Add plotter script to plot internal test's stats by Artem Titov · 5 years ago
  36. 3f17221 AEC3: Make RenderSignalAnalyzer multi-channel by Sam Zackrisson · 5 years ago
  37. b5a4ae8 Roll chromium_revision f34aba1c4b..f7cd88eb51 (696142:696270) by chromium-webrtc-autoroll · 5 years ago
  38. 1e6c415 Roll chromium_revision 783ccff90c..f34aba1c4b (696001:696142) by chromium-webrtc-autoroll · 5 years ago
  39. 087be5c Add ability to export internal state of SamplesStatsCounter. by Artem Titov · 5 years ago
  40. cc46b10 Add a usage pattern bit for host-host connections. by Qingsi Wang · 5 years ago
  41. 352b5d8 Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots. by Mirko Bonadei · 5 years ago
  42. a74e477 Deprecate legacy RtpHeaderExtensionMap::Register function by Danil Chapovalov · 5 years ago
  43. aa5a75d Embed Deceleration Target Level Offset Field Trial. by Ruslan Burakov · 5 years ago
  44. ef85f2b Clean away unused enum RtpPacketSendResult by Erik Språng · 5 years ago
  45. ca79dc6 Delete VideoReceiver2::TriggerDecoderShutdown. by Niels Möller · 5 years ago
  46. d8ac383 Delete temporary accessors in RtpDepacketizer::ParsedPayload by Danil Chapovalov · 5 years ago
  47. 3d5825e Roll chromium_revision 0d1efbbba4..783ccff90c (695897:696001) by chromium-webrtc-autoroll · 5 years ago
  48. 69f8c42 [RELAND] Add support of AudioRecord.Builder in the ADM for Android by henrika · 5 years ago
  49. dc7d2c6 Backoff to acked bitrate during first overuse detection by Per Kjellander · 5 years ago
  50. 626f7ff Update video_replay. by Sergey Silkin · 5 years ago
  51. e373bb6 Roll chromium_revision fe8ed20c77..0d1efbbba4 (695755:695897) by chromium-webrtc-autoroll · 5 years ago
  52. 9805913 Roll chromium_revision 58a2bab7bd..fe8ed20c77 (695605:695755) by chromium-webrtc-autoroll · 5 years ago
  53. a1727db Revert "Add support of AudioRecord.Builder in the ADM for Android" by Hari Molabanti · 5 years ago
  54. 7e24412 Roll chromium_revision 95ebb2b7ff..58a2bab7bd (695497:695605) by chromium-webrtc-autoroll · 5 years ago
  55. ff060ee Disable AudioDeviceTest unittests under sanitizers. by Yves Gerey · 5 years ago
  56. 0ba1705 Increase allowed jitter buffer size in ScenarioAnalyzerTest.PsnrIsLowWhenNetworkIsBad. by Jakob Ivarsson · 5 years ago
  57. 1af0f90 VP9 screenshare: use CONSTRAINED_FROM_ABOVE_DROP mode by Ilya Nikolaevskiy · 5 years ago
  58. 6fcdbc1 Store timestamp for each sample to be able to plot them in future by Artem Titov · 5 years ago
  59. 7ddea57 Add field-trial parameter to enable tests simulating a slow decoder by Johannes Kron · 5 years ago
  60. 2d7b2f5 Reland "Improve performance of RtpPacketHistory" by Erik Språng · 5 years ago
  61. 9a91161 Fixing way of printing logs because RTC_LOG() on Android has limit on printing 1024-60 characters in line. by Marin Kišić · 5 years ago
  62. 2eecfc1 Trim dependencies in modules/video_coding/ by Niels Möller · 5 years ago
  63. 16cb1f6 Stop using rtc_event.h forward header by Danil Chapovalov · 5 years ago
  64. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
  65. 47287d5 Reland "Adds peer scenario connection interface." by Sebastian Jansson · 5 years ago
  66. 70767cb Roll chromium_revision d65ce76c39..95ebb2b7ff (695395:695497) by chromium-webrtc-autoroll · 5 years ago
  67. 55f663f Roll chromium_revision b5e2f0208d..d65ce76c39 (695291:695395) by chromium-webrtc-autoroll · 5 years ago
  68. c5d2958 Roll chromium_revision 56140e7d8b..b5e2f0208d (695187:695291) by chromium-webrtc-autoroll · 5 years ago
  69. 52f7ae7 Make NetworkStateEstimator injectable in RemoteBitrateEstimator by Per Kjellander · 5 years ago
  70. 467073a Revert "Adds peer scenario connection interface." by Qingsi Wang · 5 years ago
  71. 437077d Revert "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" by Qingsi Wang · 5 years ago
  72. f6aa572 First step for introducing multichannel support for the AEC3 capture by Per Åhgren · 5 years ago
  73. 2dc1425 Roll chromium_revision a87779d34b..56140e7d8b (695071:695187) by chromium-webrtc-autoroll · 5 years ago
  74. de5f639 Removes decoder thread fallback from VideoReceiveStream. by Sebastian Jansson · 5 years ago
  75. 29ab487 Revert "Removes string support in field trial parser." by Philip Eliasson · 5 years ago
  76. 507f434 Reland "Make relative arrival delay mode default in NetEq delay manager." by Jakob Ivarsson · 5 years ago
  77. 3354157 Add support for 192kHz input audio sample rate. by henrika · 5 years ago
  78. 45b01c7 Delete some dead code in vcm::VideoReceiver and VCMReceiver by Niels Möller · 5 years ago
  79. fe407b7 Move code related to VideoCodingModule to its own build target by Niels Möller · 5 years ago
  80. 01b7e92 Mark test::DriftingClock constants as constexpr by Danil Chapovalov · 5 years ago
  81. 2486aeb Add ability to disable PSNR and SSIM computation in DVQA by Artem Titov · 5 years ago
  82. f7b1aa4 Fixing some typos. by Mirko Bonadei · 5 years ago
  83. 01e97ae Move docs about native code development into a repo directory. by Mirko Bonadei · 5 years ago
  84. 56d89da Roll chromium_revision e25e764221..a87779d34b (694813:695071) by chromium-webrtc-autoroll · 5 years ago
  85. 9bc9885 Add placeholder target to move rtc_error out of the main API target. by Mirko Bonadei · 5 years ago
  86. ee84d39 AEC3: Downmix multichannel signals before delay estimation by Gustaf Ullberg · 5 years ago
  87. d181ee7 Adds peer scenario connection interface. by Sebastian Jansson · 5 years ago
  88. 0cd61b6 MultiCodecReceiveTest: fix for flaky test. by Åsa Persson · 5 years ago
  89. b3f1487 Add ability to provide TEXT hint only when requested in PC framework by Artem Titov · 5 years ago
  90. 9509d95 Add empty build target modules/video_coding:video_coding_legacy by Niels Möller · 5 years ago
  91. c77df78 Revert "Improve performance of RtpPacketHistory" by Qingsi Wang · 5 years ago
  92. 487f9a1 Reland "Refactor SCTP data channels to use DataChannelTransportInterface." by Bjorn A Mellem · 5 years ago
  93. 116ffe7 Switch to compiling WebRTC -std=c++14 by default by Danil Chapovalov · 5 years ago
  94. a0e6ded Roll chromium_revision 75cf3925c2..e25e764221 (694706:694813) by chromium-webrtc-autoroll · 5 years ago
  95. 9e380fd Improve performance of RtpPacketHistory by Erik Språng · 5 years ago
  96. a5d952f Reland "Refactor FEC code to use COW buffers" by Ilya Nikolaevskiy · 5 years ago
  97. 4d7dac6 Remove usage of RtpRtcp::SetSSRC() in RtpRtcpImplTest by Erik Språng · 5 years ago
  98. 0987273 Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video. by Andrei Dumitru · 5 years ago
  99. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
  100. a8336d3 Connect the stable target rate to the video encoders by Florent Castelli · 5 years ago