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gerrit-public.fairphone.software / platform / external / webrtc / d3fc161c169dfe8816a5549c251481790d2cd543
  1. d3fc161 Revert "Change buffer level filter to store current level in number of samples." by Jakob Ivarsson · 6 years ago
  2. 87977dd Change buffer level filter to store current level in number of samples. by Jakob Ivarsson · 6 years ago
  3. 3391072 Use DefaultTaskQueueFactory in CreatePeerConnectionFactoryForJava by Danil Chapovalov · 6 years ago
  4. a95949e Switch rtp_generator to ABSL_FLAG. by Mirko Bonadei · 6 years ago
  5. beff535 Switch frame_editor to ABSL_FLAG. by Mirko Bonadei · 6 years ago
  6. 6d6b0fb Switch reference_less_video_analysis to ABSL_FLAG. by Mirko Bonadei · 6 years ago
  7. 8e032c8 Switch psnr_ssim_analyzer to ABSL_FLAG. by Mirko Bonadei · 6 years ago
  8. 2308bb2 Switch rgba_to_i420_converter to ABSL_FLAG. by Mirko Bonadei · 6 years ago
  9. 6bafc8b Reland "Switch frame_analyzer to ABSL_FLAG." by Mirko Bonadei · 6 years ago
  10. 90c6d1b Revert "Switch frame_analyzer to ABSL_FLAG." by Mirko Bonadei · 6 years ago
  11. 2923d76 Roll chromium_revision c049034cc7..6c5bbd86c3 (671848:671957) by chromium-webrtc-autoroll · 6 years ago
  12. 3668e9b Switch frame_analyzer to ABSL_FLAG. by Mirko Bonadei · 6 years ago
  13. fa1e7c1 Roll chromium_revision d6c73f3029..c049034cc7 (671737:671848) by chromium-webrtc-autoroll · 6 years ago
  14. f4996d4 Roll chromium_revision 163d6b201e..d6c73f3029 (671624:671737) by chromium-webrtc-autoroll · 6 years ago
  15. e9cac4f Improvements to scenario video stats for scenario tests. by Sebastian Jansson · 6 years ago
  16. 8605fbf [getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc. by Henrik Boström · 6 years ago
  17. 8f7b41a Roll chromium_revision ec99dbf1aa..163d6b201e (671513:671624) by chromium-webrtc-autoroll · 6 years ago
  18. 4b8a5b4 Removes unused PacketFeedbackComparator by Sebastian Jansson · 6 years ago
  19. f48bca7 Avoid triggering a false error logging when using encryptor and sending DTX. by Minyue Li · 6 years ago
  20. 59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 6 years ago
  21. 13eb764 Move towards always using packet type instead of priority in RTPSender by Erik Språng · 6 years ago
  22. 1231419 BalancedDegradationSettings: Add option to configure QP thresholds. by Åsa Persson · 6 years ago
  23. 88290ae Reland "Cleanup of RTP references in GoogCC implementation." by Sebastian Jansson · 6 years ago
  24. 746dd0d Add support for two-byte RTP header extensions to UsedRtpHeaderExtensionIds by Johannes Kron · 6 years ago
  25. 2ffcd82 Make DefaultTemporalLayers explicitly request a key frame by Elad Alon · 6 years ago
  26. 9aa870a Fixing fuzzer by backing up and restoring `packet_info`. by Chen Xing · 6 years ago
  27. 214f543 Make useful padding the default. by Erik Språng · 6 years ago
  28. 2b1d8c8 Roll chromium_revision b827eec031..ec99dbf1aa (671413:671513) by chromium-webrtc-autoroll · 6 years ago
  29. 4a15833 Roll chromium_revision a8f7a25b6e..b827eec031 (671303:671413) by chromium-webrtc-autoroll · 6 years ago
  30. 3e3a900 Roll chromium_revision ffeecb2d27..a8f7a25b6e (671198:671303) by chromium-webrtc-autoroll · 6 years ago
  31. abf7eba Roll chromium_revision ef202aa763..ffeecb2d27 (671098:671198) by chromium-webrtc-autoroll · 6 years ago
  32. 6fd5b9e Roll chromium_revision 568fb40139..ef202aa763 (670985:671098) by chromium-webrtc-autoroll · 6 years ago
  33. 6e91646 Roll chromium_revision 5cf576bfbe..568fb40139 (670869:670985) by chromium-webrtc-autoroll · 6 years ago
  34. 6924d50 Roll chromium_revision 4dfb50605a..5cf576bfbe (670751:670869) by chromium-webrtc-autoroll · 6 years ago
  35. d487a55 Revert "Remove sync buffer length from FilteredCurrentDelayMs." by Jakob Ivarsson · 6 years ago
  36. a550dad Minor rtc_event_log_impl cleanup. by Nikita Zetilov · 6 years ago
  37. 6c42d92 Added video_coding::EncodedFrame copy ctor. by philipel · 6 years ago
  38. f00bf42 Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker. by Chen Xing · 6 years ago
  39. 7953ad5 Revert "Cleanup of RTP references in GoogCC implementation." by Sebastian Jansson · 6 years ago
  40. fa79081 Cleanup of RTP references in GoogCC implementation. by Sebastian Jansson · 6 years ago
  41. 775c02e Do not use libevent when targeting wasm. by Mirko Bonadei · 6 years ago
  42. 9da25bd In PeerConnection unittests set TaskQueueFactory explicitly by Danil Chapovalov · 6 years ago
  43. 621be83 Roll chromium_revision 516b926bdf..4dfb50605a (670612:670751) by chromium-webrtc-autoroll · 6 years ago
  44. e3cc489 Add logging and edit the field trial name for piggyback ICE check by Qingsi Wang · 6 years ago
  45. 0d92f15 Roll chromium_revision 39e46dcf9e..516b926bdf (670459:670612) by chromium-webrtc-autoroll · 6 years ago
  46. 5bf5806 Force DefaultTaskQueueFactory in legacy CreatePeerConnectionFactory by Danil Chapovalov · 6 years ago
  47. c2a54dc Roll chromium_revision 6e638caa96..39e46dcf9e (670359:670459) by chromium-webrtc-autoroll · 6 years ago
  48. c16289f Split the build rule for video_frame into a video_rtp_headers part. by Chen Xing · 6 years ago
  49. b64ad0e Using Clock::CurrentTime() where non-test behavior is unchanged. by Sebastian Jansson · 6 years ago
  50. 18f1f0c Revert "Raise IllegalStateException for calls to retain() or release() on zero ref count" by Niels Moller · 6 years ago
  51. 8a959bf Raise IllegalStateException for calls to retain() or release() on zero ref count by Niels Möller · 6 years ago
  52. 505bac2 Add default implementation of deprecated StartAecDump method. by Niels Möller · 6 years ago
  53. 4d504c7 New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer by Niels Möller · 6 years ago
  54. 0894f0f Add piggyback acknowledgement of the last ICE check received in outgoing checks. by Qingsi Wang · 6 years ago
  55. f9511fc Roll chromium_revision 619b073720..6e638caa96 (670258:670359) by chromium-webrtc-autoroll · 6 years ago
  56. b7a3e3f Roll chromium_revision 8639acbee7..619b073720 (670130:670258) by chromium-webrtc-autoroll · 6 years ago
  57. e8347a8 Roll chromium_revision bf62d746a4..8639acbee7 (669828:670130) by chromium-webrtc-autoroll · 6 years ago
  58. 1b3f4f9 Allow RtpPacketHistory encapsulator function to abort retransmit by Erik Språng · 6 years ago
  59. 4cbb4ef Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest by Oleh Prypin · 6 years ago
  60. b762b5b Fix potential signed overflow in IntervalBudget::set_target_rate_kbps by Per Kjellander · 6 years ago
  61. 342f98b Fixes for flexfec crash in scenario tests. by Sebastian Jansson · 6 years ago
  62. 58ee187 Add support within PacedSender and pacer queue for owning rtp packets. by Erik Språng · 6 years ago
  63. b028c6a Support __EMSCRIPTEN__ in rtc_base. by Mirko Bonadei · 6 years ago
  64. 0c0c969 Add/rewrite H264 VUI video signal type description. by Sergey Silkin · 6 years ago
  65. 449888e Cleanup of resources from removed remote bitrate estimate test framework. by Sebastian Jansson · 6 years ago
  66. 2c648f5 Stop running 'bwe_simulations_tests'. by Mirko Bonadei · 6 years ago
  67. e181440 Fix documentation in BitrateAdjuster. by Sami Kalliomäki · 6 years ago
  68. 0f557fe Removes unused dependency on RTP/RTCP from loss based controller. by Sebastian Jansson · 6 years ago
  69. 61d8ee1 Roll chromium_revision 2ce8c83798..6ae0f0cd4c (669595:669703) by chromium-webrtc-autoroll · 6 years ago
  70. 23026ee Adds SortedByReceiveTime to TransportPacketsFeedback. by Sebastian Jansson · 6 years ago
  71. 873a7a9 Fix event_log_visualizer help text and default profile. by Bjorn Terelius · 6 years ago
  72. 9c771c2 Add TrySendPacket() method to RTP modules. by Erik Språng · 6 years ago
  73. d9c900f Add Clone() to Vp8FrameBufferControllerFactory by Elad Alon · 6 years ago
  74. 1dee91a Roll chromium_revision d958b08217..2ce8c83798 (669491:669595) by chromium-webrtc-autoroll · 6 years ago
  75. e93454a Removes AddAndRemoveOld from SendTimeHistory by Sebastian Jansson · 6 years ago
  76. 7b99fa8 Roll chromium_revision 447b80d261..d958b08217 (669391:669491) by chromium-webrtc-autoroll · 6 years ago
  77. 2f42dab Roll chromium_revision e5d3b43486..447b80d261 (669285:669391) by chromium-webrtc-autoroll · 6 years ago
  78. ef5e35d Roll chromium_revision be90998a40..e5d3b43486 (669184:669285) by chromium-webrtc-autoroll · 6 years ago
  79. e112bb8 Adds support for abs send time extension in scenario tests. by Sebastian Jansson · 6 years ago
  80. 3d61ab1 Adds send time to ReceivedPacket struct. by Sebastian Jansson · 6 years ago
  81. 8aba8fe Reland "Populate the GFD-00 for H264 and generic codecs." by philipel · 6 years ago
  82. bdb6b39 Let HardwareVideoEncoder cache result from codec.getOutputBuffers() by Niels Möller · 6 years ago
  83. 5e953d7 Insert startcodes for H264 single NALU packets. by philipel · 6 years ago
  84. 856ca19 Delete unused method ReportBlockStats::AggregateAndStore by Niels Möller · 6 years ago
  85. 0292e97 Roll chromium_revision f165a6d5de..be90998a40 (669076:669184) by chromium-webrtc-autoroll · 6 years ago
  86. 607a6f1 Moves conversion to ReceivedPacket from RtpPacketReceived to Call. by Sebastian Jansson · 6 years ago
  87. 98cbb22 Moved AsyncInvoker to be destructed first in WebRtcVideoSendStream. by philipel · 6 years ago
  88. 7f2a67f Cleanup names and extra checks for errors in PC test framework by Artem Titov · 6 years ago
  89. 12d64de Remove sequence_number from RtpPacketInfo. by Chen Xing · 6 years ago
  90. ad82e8e Fix: restore disabling PC smoke test on iOS by Artem Titov · 6 years ago
  91. 19a1d50 Refactor WavWriter to use FileWrapper rather than PlatformFile by Niels Möller · 6 years ago
  92. 04e129a Revert "Populate the GFD-00 for H264 and generic codecs." by Philip Eliasson · 6 years ago
  93. 1a285e0 Roll chromium_revision 96eca2d491..f165a6d5de (668951:669076) by chromium-webrtc-autoroll · 6 years ago
  94. a9a2a75 Roll chromium_revision 22f062d604..96eca2d491 (668845:668951) by chromium-webrtc-autoroll · 6 years ago
  95. c5effc2 Remove DataContentDescription shim by Harald Alvestrand · 6 years ago
  96. ef3fd9c Add support for simulcast with Vp8 from caller into PC level quality tests. by Artem Titov · 6 years ago
  97. 6751260 Roll chromium_revision b08bd9b643..22f062d604 (668716:668845) by chromium-webrtc-autoroll · 6 years ago
  98. d3c6f9c Populate the GFD-00 for H264 and generic codecs. by philipel · 6 years ago
  99. 9e25f74 Update visibility for JNI targets in sdk/android. by Sami Kalliomäki · 6 years ago
  100. 5894b6a Add kPayloadTypeGeneric to CallTest and use it in VideoQualityTest. by Rasmus Brandt · 6 years ago