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gerrit-public.fairphone.software / platform / external / webrtc / 7f57788ab7e2f9ee03086ab3c4a14c1ad18b49e6
  1. 7f57788 Removes trial to enable BBR congestion controller. by Sebastian Jansson · 6 years ago
  2. e2e0000 Make pacing buffer send interval configurable. by Christoffer Rodbro · 6 years ago
  3. 61832dd Propagate spatial index to EncodedImage. by Sergey Silkin · 6 years ago
  4. 76fd7b4 Roll chromium_revision bdf260ef72..5397835a49 (617896:618202) by chromium-webrtc-autoroll · 6 years ago
  5. 3793bb4 Refactor TestVideoCapturer to support multiple sinks. by Niels Möller · 6 years ago
  6. 941fd43 Update number of spatial layers if SS is available. by Sergey Silkin · 6 years ago
  7. 2894750 Adds flag to customize output root dir in scenario tests. by Sebastian Jansson · 6 years ago
  8. d69998a Adds support for empty key fields in field trial parser. by Sebastian Jansson · 6 years ago
  9. afa07dd [Unified Plan] SRD: Always set associated remote streams. by Henrik Boström · 6 years ago
  10. 02c4f15 Stop using deprecated PacedSender method from RtpTransportControllerSend. by Sebastian Jansson · 6 years ago
  11. 29a935a Refactor NetEqDecoderPlc to use AudioDecoderProxyFactory by Niels Möller · 6 years ago
  12. f65309c Removes return value and Try prefix from TryDeliverPacket. by Sebastian Jansson · 6 years ago
  13. 9a4f38e Adds optional video quality metrics to scenario tests. by Sebastian Jansson · 6 years ago
  14. 54fa024 Removed log from StatsCollector::GetTrackIdBySsrc. by Seth Hampson · 6 years ago
  15. ceac015 Fail SetLocalDescription if a=mid lines are missing by Steve Anton · 6 years ago
  16. b275788 Register stat callbacks after rate observer is registered. by Piotr (Peter) Slatala · 6 years ago
  17. 322bba7 Roll chromium_revision 7ebbeeebe7..bdf260ef72 (617780:617896) by chromium-webrtc-autoroll · 6 years ago
  18. e920351 Update PeerConnectionProxy to reflect new PeerConnectionInterface methods by Steve Anton · 6 years ago
  19. f5fc537 Allow H264 simulcast be default by Erik Språng · 6 years ago
  20. 3f651d8 Reland "Add AudioDecoderFactory to NetEqTest constructor." by Niels Möller · 6 years ago
  21. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  22. aabac23 Roll chromium_revision 638370e07d..7ebbeeebe7 (617680:617780) by chromium-webrtc-autoroll · 6 years ago
  23. f2090c1 Roll chromium_revision eaca19bda5..638370e07d (617557:617680) by chromium-webrtc-autoroll · 6 years ago
  24. 06817cd [Unified Plan] Support legacy endpoints that do not use a=mid by Steve Anton · 6 years ago
  25. d1208c2 Desktop capturer: Add OnDisplayChanged callback by Gary Kacmarcik · 6 years ago
  26. 5a1de87 Use unique_ptr in webrtcsdp.cc by Steve Anton · 6 years ago
  27. c26337d Roll chromium_revision 640d842b5c..eaca19bda5 (617449:617557) by chromium-webrtc-autoroll · 6 years ago
  28. 503da94 Adds first unacknowledged packet send time. by Sebastian Jansson · 6 years ago
  29. 11b8703 Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs by Sam Zackrisson · 6 years ago
  30. 358fba1 Removes NetworkControllerTester by Sebastian Jansson · 6 years ago
  31. 1c931c4 Use VideoSourceInterface for owning test video sources by Niels Möller · 6 years ago
  32. f0d0312 Allows injection of network controller in scenarios. by Sebastian Jansson · 6 years ago
  33. b3564c1 Back off relative to current estimate rather than ack rate when in ALR. by Erik Språng · 6 years ago
  34. a1bec23 Pass on explicit color space for VP8 and H264 by Johannes Kron · 6 years ago
  35. 0fcf4b1 Delete unused I420 "codec" by Niels Möller · 6 years ago
  36. e1190c6 Roll chromium_revision 56612bfb38..640d842b5c (617342:617449) by chromium-webrtc-autoroll · 6 years ago
  37. 1699408 Inlining IncomingVideoStream::NewFrameTask. by Sebastian Jansson · 6 years ago
  38. 7d92de6 Deprecating legacy SendSideCongestionController. by Sebastian Jansson · 6 years ago
  39. 4415315 Move ServerSocket code to separate files and into test target by Niels Möller · 6 years ago
  40. 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
  41. 8eeccbe Delete Start and Stop methods from TestVideoCapturer. by Niels Möller · 6 years ago
  42. 41f3a43 Remove CodecInst pt.3 by Fredrik Solenberg · 6 years ago
  43. de133ce Roll chromium_revision caa5e0d1c9..56612bfb38 (617225:617342) by chromium-webrtc-autoroll · 6 years ago
  44. 7a33d23 Roll chromium_revision ddce7e097d..caa5e0d1c9 (617111:617225) by chromium-webrtc-autoroll · 6 years ago
  45. 7d8c27e [Fuchsia] Implement detection of available cores. by Fabrice de Gans-Riberi · 6 years ago
  46. 68586e8 Replace starts_with and ends_with with Abseil by Steve Anton · 6 years ago
  47. 73f2da9 Remove VP8EncoderSimulcastProxy by Florent Castelli · 6 years ago
  48. f7f753b Revert "Add AudioDecoderFactory to NetEqTest constructor." by Oleh Prypin · 6 years ago
  49. 74a99e7 Roll chromium_revision 380d56dda2..ddce7e097d (617001:617111) by chromium-webrtc-autoroll · 6 years ago
  50. 3ff71de Android: Add option to mirror vertically in EglRenderer by Magnus Jedvert · 6 years ago
  51. eb02ecd Move peerconnectionwrapper.(h|cc) into separate build target by Artem Titov · 6 years ago
  52. daa970f Add AudioDecoderFactory to NetEqTest constructor. by Niels Möller · 6 years ago
  53. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  54. a134204 Reland "Move relay server code to a test-only target p2p_server_utils." by Niels Möller · 6 years ago
  55. 194d4d2 Delete unused send-side methods of VideoCodingModule by Niels Möller · 6 years ago
  56. 2d8b60c Roll chromium_revision 31225b9c5f..380d56dda2 (616901:617001) by chromium-webrtc-autoroll · 6 years ago
  57. adae52b Roll chromium_revision 84fedb7e68..31225b9c5f (616714:616901) by chromium-webrtc-autoroll · 6 years ago
  58. e4e5273 Roll chromium_revision 6d37479eb8..84fedb7e68 (616603:616714) by chromium-webrtc-autoroll · 6 years ago
  59. 2db46b0 Added new feature to print a text log to neteq_rtpplay by Ivo Creusen · 6 years ago
  60. 5bb1afd Revert "Switch to logdog binary provided by IMPLIED_CIPD_BINARIES" by Oleh Prypin · 6 years ago
  61. 94f1074 Only use GetAudio events that correspond to an ssrc matching at least one incoming packet. by Ivo Creusen · 6 years ago
  62. 24779d8 Missing packet send time should not cause BWE backoff. by Bjorn Terelius · 6 years ago
  63. e05d720 Switch to logdog binary provided by IMPLIED_CIPD_BINARIES by Oleh Prypin · 6 years ago
  64. 3073f3d Revert "Reland "Default to dlopening the PipeWire."" by Henrik Grunell · 6 years ago
  65. 3be607f Use output_dir instead of output_name by Takuto Ikuta · 6 years ago
  66. f1ab9b9 Refactor creation of ColorSpace test data by Johannes Kron · 6 years ago
  67. 25aefd3 Delete log severity LS_SENSITIVE by Niels Möller · 6 years ago
  68. 5301951 Roll chromium_revision 55b877610b..6d37479eb8 (615952:616603) by chromium-webrtc-autoroll · 6 years ago
  69. 659dfa7 update luci-go client by Takuto Ikuta · 6 years ago
  70. 944c755 Use unique_ptr in DataChannel PacketQueue by Steve Anton · 6 years ago
  71. cca13f6 Remove unused cryptoparams.h header by Steve Anton · 6 years ago
  72. 9e2e6e7 Set LoggedIceCandidatePairEvent.transaction_id to default value if missing. by Zach Stein · 6 years ago
  73. 3e94557 Adding fuzzing configuration files for Rtp Replay Fuzzing. by Benjamin Wright · 6 years ago
  74. 0cc42d4 Reland "Default to dlopening the PipeWire." by Tomas Popela · 6 years ago
  75. 833979f Adding metrics for hostname candidate use. by Jeroen de Borst · 6 years ago
  76. fe79b34 Reorder methods and members of HdrMetadata by Johannes Kron · 6 years ago
  77. d96b275 Refactor EncodeParameters usage, remove unused rtt/loss by Erik Språng · 6 years ago
  78. e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
  79. 94c0f26 Android: One weird trick for avoiding graphics deadlocks by Magnus Jedvert · 6 years ago
  80. 2aa0379 Increase timeout of webrtc_perf_tests on Android to 1h15m by Oleh Prypin · 6 years ago
  81. aa7bc7e Create field trial for vp8 number of thread on iOS. by Jiawei Ou · 6 years ago
  82. 1dac6d8 Sanitize candidates in ICE-level stats when necessary. by Qingsi Wang · 6 years ago
  83. cb3ad28 Roll chromium_revision 23962c3887..55b877610b (615838:615952) by chromium-webrtc-autoroll · 6 years ago
  84. 1618095 Cleanup of RtpTransportControllerSend. by Sebastian Jansson · 6 years ago
  85. 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
  86. f7f13e0 Add end-to-end test for ColorSpace information by Johannes Kron · 6 years ago
  87. 3d5df13 Switch to literals in playout delay tests by Paulina Hensman · 6 years ago
  88. c13f4be Add chroma siting to color space RTP extension by Johannes Kron · 6 years ago
  89. 0697ce2 Revert "Reland "Default to dlopening the PipeWire."" by Henrik Grunell · 6 years ago
  90. 1d8307d Delete VideoCodec::targetBitrate by Niels Möller · 6 years ago
  91. ab64e8a Using fully qualified names for using declarations. by Mirko Bonadei · 6 years ago
  92. 17d57c7 Reintroduce division by two for audio playout delay by Paulina Hensman · 6 years ago
  93. e10b163 Stop using 'using namespace'. by Mirko Bonadei · 6 years ago
  94. cd39297 Roll chromium_revision 7f93585b0f..23962c3887 (615733:615838) by chromium-webrtc-autoroll · 6 years ago
  95. 2701bc9 Signals start rate when registering to TargetTransferRateObserver. by Sebastian Jansson · 6 years ago
  96. 50b66d5 Convert NetEq Cng-related test to not use RegisterExternalDecoder by Niels Möller · 6 years ago
  97. a099877 Reland "Default to dlopening the PipeWire." by Tomas Popela · 6 years ago
  98. f6b10fb Basic fuzzing of rtc::s_url_decode. by Benjamin Wright · 6 years ago
  99. 60446b8 Roll chromium_revision 84fd8a4f89..7f93585b0f (615612:615733) by chromium-webrtc-autoroll · 6 years ago
  100. c57d573 RID parsing for Simulcast support. by Amit Hilbuch · 6 years ago