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gerrit-public.fairphone.software / platform / external / webrtc / f288c8ee6e4d1e948f946f8027eb4262bf291967
  1. f288c8e Roll chromium_revision cf1a2beb4b..fc1e948f93 (697976:698112) by chromium-webrtc-autoroll · 5 years ago
  2. c12db81 Add frame receive to frame rendered metric to video_quality_analyzer by Johannes Kron · 5 years ago
  3. f0be5b5 Make GetBitstream non-virtual since it is no longer needed for testing. by philipel · 5 years ago
  4. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  5. ac315b2 Add support for max/min encode bitrate to peer connection quality test by Johannes Kron · 5 years ago
  6. 6a09263 Delete obsolete isac "assign" api by Niels Möller · 5 years ago
  7. d8ffbb0 Roll chromium_revision afdb2e7a8b..cf1a2beb4b (697871:697976) by chromium-webrtc-autoroll · 5 years ago
  8. 76161f7 Move the call to GetBitstream out of the RtpFrameObject ctor. by philipel · 5 years ago
  9. 14137a1 Adds logging of audio sessions status on the recording side in ADM for Android. by henrika · 5 years ago
  10. 86873f0 Improve field trial error message. by Björn Terelius · 5 years ago
  11. e942b14 New build target api:media_interface by Niels Möller · 5 years ago
  12. 0a5ed89 Adds remote estimates to rtc event log. by Sebastian Jansson · 5 years ago
  13. 6ed60e3 Implement Dependency Descriptor writer by Danil Chapovalov · 5 years ago
  14. 489843f Improve trendline estimator logging. by Björn Terelius · 5 years ago
  15. 693bf1e Delete modules/rtp_rtcp local DivideRoundToNearest in favor on one in rtc_base by Danil Chapovalov · 5 years ago
  16. bd24260 Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871) by chromium-webrtc-autoroll · 5 years ago
  17. efa04ef Roll chromium_revision 65274319fc..eae7ecf757 (697640:697744) by chromium-webrtc-autoroll · 5 years ago
  18. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  19. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
  20. 4bad650 Roll chromium_revision 2bd75c72c1..65274319fc (697505:697640) by chromium-webrtc-autoroll · 5 years ago
  21. 7b04a91 Delete almost all default methods on PeerConnectionInterface by Niels Möller · 5 years ago
  22. e607a06 Removed unused include from PacketBuffer. by philipel · 5 years ago
  23. 33b83fd Introduce integer division helpers with non-default rounding by Danil Chapovalov · 5 years ago
  24. b6a45dd Revert "Fix minor regression caused by a8336d3" by Evan Shrubsole · 5 years ago
  25. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
  26. 1162ba2 Add max/min encode bitrates to video config of peer connection tests by Johannes Kron · 5 years ago
  27. 7cfde54 Roll chromium_revision 51a0808947..2bd75c72c1 (697405:697505) by chromium-webrtc-autoroll · 5 years ago
  28. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 5 years ago
  29. c128df1 Update style guide for absl::make_unique. by Mirko Bonadei · 5 years ago
  30. 95c4b91 Roll chromium_revision 31d9542abc..51a0808947 (697288:697405) by chromium-webrtc-autoroll · 5 years ago
  31. ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 5 years ago
  32. 4d461ba Reusing MediaStreamAllocationConfig struct in ObserverConfig. by Sebastian Jansson · 5 years ago
  33. 86314cf Cleaning up C++14 move into lambda TODOs. by Sebastian Jansson · 5 years ago
  34. 368d002 Roll chromium_revision dbd1569418..31d9542abc (697157:697288) by chromium-webrtc-autoroll · 5 years ago
  35. 9fa8ef1 absl::make_unique presubmit check. by Mirko Bonadei · 5 years ago
  36. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  37. 809198e Fix minor regression caused by a8336d3 by Evan Shrubsole · 5 years ago
  38. 7d00342 Remove old packet socket factory header. by Patrik Höglund · 5 years ago
  39. e1b7777 Removing deprecated min_pacing_rate alias in StreamsConfig. by Sebastian Jansson · 5 years ago
  40. 4a822f4 Roll chromium_revision 2e4ccff8a8..dbd1569418 (696956:697157) by chromium-webrtc-autoroll · 5 years ago
  41. 2c6ea52 In TaskQueueTest::PostDelayedAfterDesctruct increase timeout by Danil Chapovalov · 5 years ago
  42. c1c6284 New (empty) build target api:media_stream_interface by Niels Möller · 5 years ago
  43. 1722182 Roll chromium_revision 3cf04dec00..2e4ccff8a8 (696812:696956) by chromium-webrtc-autoroll · 5 years ago
  44. 7262fc2 Refactor Rtp Receivers to accept SSRC 0. by Saurav Das · 5 years ago
  45. 3d16474 in RtcpTransciever use lambdas with move capture. by Danil Chapovalov · 5 years ago
  46. 3462793 Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812) by chromium-webrtc-autoroll · 5 years ago
  47. 68ef259 Delete deprecated rtc_event.h file by Danil Chapovalov · 5 years ago
  48. f5dec1c Implement Dependency Descriptor reader by Danil Chapovalov · 5 years ago
  49. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 5 years ago
  50. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 5 years ago
  51. 1b6a30d Update WebRTC's C++ style guide to reflect the switch to C++14. by Mirko Bonadei · 5 years ago
  52. a740142 Refactor LossNotificationController to not use VCMPacket by Niels Möller · 5 years ago
  53. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
  54. c4e80ad Delete forward declarations from peer_connection_interface.h by Niels Möller · 5 years ago
  55. 7af1bb3 Roll chromium_revision 9f15168729..1d12ff693d (696593:696696) by chromium-webrtc-autoroll · 5 years ago
  56. fcbe407 Adding more refined control over choice of band-splitting by Per Åhgren · 5 years ago
  57. ec06ebd Roll chromium_revision 9004bcf36a..9f15168729 (696490:696593) by chromium-webrtc-autoroll · 5 years ago
  58. 0dd37ce Roll chromium_revision 4740202690..9004bcf36a (696373:696490) by chromium-webrtc-autoroll · 5 years ago
  59. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  60. 6a6eb61 Roll chromium_revision f7cd88eb51..4740202690 (696270:696373) by chromium-webrtc-autoroll · 5 years ago
  61. e78fd80 New class DummyPeerConnection by Niels Möller · 5 years ago
  62. 3873927 Fix time units in plotted charts by Artem Titov · 5 years ago
  63. 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 5 years ago
  64. 0a7d5d8 Set console window NOTOPMOST flag after WindowFinderTest.FindDrawerWindow on Windows by Kimmo Kinnunen · 5 years ago
  65. 01be33b Using lambdas instead of rtc::Bind in BaseChannel. by Sebastian Jansson · 5 years ago
  66. 262bbae Fix rare audioLevel flake in RTCStatsIntegrationTest. by Henrik Boström · 5 years ago
  67. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  68. 5f15f86 Add plotter script to plot internal test's stats by Artem Titov · 5 years ago
  69. 3f17221 AEC3: Make RenderSignalAnalyzer multi-channel by Sam Zackrisson · 5 years ago
  70. b5a4ae8 Roll chromium_revision f34aba1c4b..f7cd88eb51 (696142:696270) by chromium-webrtc-autoroll · 5 years ago
  71. 1e6c415 Roll chromium_revision 783ccff90c..f34aba1c4b (696001:696142) by chromium-webrtc-autoroll · 5 years ago
  72. 087be5c Add ability to export internal state of SamplesStatsCounter. by Artem Titov · 5 years ago
  73. cc46b10 Add a usage pattern bit for host-host connections. by Qingsi Wang · 5 years ago
  74. 352b5d8 Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots. by Mirko Bonadei · 5 years ago
  75. a74e477 Deprecate legacy RtpHeaderExtensionMap::Register function by Danil Chapovalov · 5 years ago
  76. aa5a75d Embed Deceleration Target Level Offset Field Trial. by Ruslan Burakov · 5 years ago
  77. ef85f2b Clean away unused enum RtpPacketSendResult by Erik Språng · 5 years ago
  78. ca79dc6 Delete VideoReceiver2::TriggerDecoderShutdown. by Niels Möller · 5 years ago
  79. d8ac383 Delete temporary accessors in RtpDepacketizer::ParsedPayload by Danil Chapovalov · 5 years ago
  80. 3d5825e Roll chromium_revision 0d1efbbba4..783ccff90c (695897:696001) by chromium-webrtc-autoroll · 5 years ago
  81. 69f8c42 [RELAND] Add support of AudioRecord.Builder in the ADM for Android by henrika · 5 years ago
  82. dc7d2c6 Backoff to acked bitrate during first overuse detection by Per Kjellander · 5 years ago
  83. 626f7ff Update video_replay. by Sergey Silkin · 5 years ago
  84. e373bb6 Roll chromium_revision fe8ed20c77..0d1efbbba4 (695755:695897) by chromium-webrtc-autoroll · 5 years ago
  85. 9805913 Roll chromium_revision 58a2bab7bd..fe8ed20c77 (695605:695755) by chromium-webrtc-autoroll · 5 years ago
  86. a1727db Revert "Add support of AudioRecord.Builder in the ADM for Android" by Hari Molabanti · 5 years ago
  87. 7e24412 Roll chromium_revision 95ebb2b7ff..58a2bab7bd (695497:695605) by chromium-webrtc-autoroll · 5 years ago
  88. ff060ee Disable AudioDeviceTest unittests under sanitizers. by Yves Gerey · 5 years ago
  89. 0ba1705 Increase allowed jitter buffer size in ScenarioAnalyzerTest.PsnrIsLowWhenNetworkIsBad. by Jakob Ivarsson · 5 years ago
  90. 1af0f90 VP9 screenshare: use CONSTRAINED_FROM_ABOVE_DROP mode by Ilya Nikolaevskiy · 5 years ago
  91. 6fcdbc1 Store timestamp for each sample to be able to plot them in future by Artem Titov · 5 years ago
  92. 7ddea57 Add field-trial parameter to enable tests simulating a slow decoder by Johannes Kron · 5 years ago
  93. 2d7b2f5 Reland "Improve performance of RtpPacketHistory" by Erik Språng · 5 years ago
  94. 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
  95. 2eecfc1 Trim dependencies in modules/video_coding/ by Niels Möller · 5 years ago
  96. 16cb1f6 Stop using rtc_event.h forward header by Danil Chapovalov · 5 years ago
  97. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
  98. 47287d5 Reland "Adds peer scenario connection interface." by Sebastian Jansson · 5 years ago
  99. 70767cb Roll chromium_revision d65ce76c39..95ebb2b7ff (695395:695497) by chromium-webrtc-autoroll · 5 years ago
  100. 55f663f Roll chromium_revision b5e2f0208d..d65ce76c39 (695291:695395) by chromium-webrtc-autoroll · 5 years ago