Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 12dc1842d62ee8df1e462f9b6a617fef9ab8b3b7
  1. 12dc184 Reland "Some cleanup for the logging code:" by Tommi · 7 years ago
  2. 3359bd1 Roll chromium_revision b6ed4eb6cc..dba3f95458 (539140:539260) by Autoroller · 7 years ago
  3. 72eeaa3 Revert "Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller." by Zhi Huang · 7 years ago
  4. 9f79a92 Roll chromium_revision 3768946207..b6ed4eb6cc (539034:539140) by Autoroller · 7 years ago
  5. 278aa42 Revert "Some cleanup for the logging code:" by Philip Eliasson · 7 years ago
  6. 9e24cb3 Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these. by Dino Radaković · 7 years ago
  7. bd7b461 Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller. by Alex Loiko · 7 years ago
  8. 9ecdcdf Some cleanup for the logging code: by Tommi · 7 years ago
  9. 151be2d comfort_noise_decoder_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
  10. 06fa153 neteq_rtp_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
  11. 2a6d864 neteq_signal_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
  12. 8b84365 NetEq: Guarding against reading outside of memory by Henrik Lundin · 7 years ago
  13. 132e28e Add thread checks to ReceiveStatisticsProxy that reflect design comments. by Tommi · 7 years ago
  14. fdf50a6 Roll chromium_revision 29f6e2e8b6..3768946207 (538933:539034) by Autoroller · 7 years ago
  15. c392866 Implement certificate chain stats. by Taylor Brandstetter · 7 years ago
  16. 29ef9f0 Roll chromium_revision 1cf758f803..29f6e2e8b6 (538618:538933) by Autoroller · 7 years ago
  17. 88f6dec Multiplex Codec Bug Fix: Padding Needed For H264 by Qiang Chen · 7 years ago
  18. 1807d57 Add application_data field(s) to RtpPacketToSend and PacketOptions. by Dino Radaković · 7 years ago
  19. f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
  20. 225c787 Move default thresholds from QualityScaler to encoders. by Niels Möller · 7 years ago
  21. 12fb170 Added some margin to ramp down target in perf test. by Sebastian Jansson · 7 years ago
  22. 93db0d8 Whitespace change by Oleh Prypin · 7 years ago
  23. d60d5c4 Improve Java video codec error handling. by Sami Kalliomäki · 7 years ago
  24. 99f52f8 Make decoder software fallback sticky. by Sami Kalliomäki · 7 years ago
  25. a2e3ab1 Conditionally include real_fourier_openmax.h. by Mirko Bonadei · 7 years ago
  26. c33c0fc Moved pacer and congestion thread from call. by Sebastian Jansson · 7 years ago
  27. 604ab19 Roll chromium_revision eb957d794e..1cf758f803 (538454:538618) by Autoroller · 7 years ago
  28. e818b6e Create the JsepTransportController and JsepTransport2. by Zhi Huang · 7 years ago
  29. fbf3bce Reland "Reduce locking in VideoReceiver and check the threading model." by Tommi · 7 years ago
  30. ee52562 Roll chromium_revision 37c4da4be1..eb957d794e (538199:538454) by Autoroller · 7 years ago
  31. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  32. 89c7938 Delete assumption TimeMicrosToNtp can match RealTimeClock by Danil Chapovalov · 7 years ago
  33. 6ce0359 Adding missing ASM dependencies. by Mirko Bonadei · 7 years ago
  34. e7c891f Renamed FrameObject to EncodedFrame. by philipel · 7 years ago
  35. a5c735f Fixed observer unsubscribtion in RTCRtpReceiver. by Yura Yaroshevich · 7 years ago
  36. c668108 Maintain audio receive stream gain across recreations by Oskar Sundbom · 7 years ago
  37. 415920b Return correct subtype from RTCRtpSender/Receiver track. by Yura Yaroshevich · 7 years ago
  38. 35dd6cd Added dependencies to mock transport controller send. by Sebastian Jansson · 7 years ago
  39. 3f06c3b Change text output from VideoProcessor slightly. by Rasmus Brandt · 7 years ago
  40. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  41. dfde334 Adding SendSideCongestionControllerInterface. by Sebastian Jansson · 7 years ago
  42. 0404225 ClosePlatformFile() on non-Windows: Return true on success, false on failure by Karl Wiberg · 7 years ago
  43. 518716f Delete left-over declarations. by Niels Möller · 7 years ago
  44. 64cf731 Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199) by Mirko Bonadei · 7 years ago
  45. 9a03dd8 Removed new calls on RtpTransportControllerSend. by Sebastian Jansson · 7 years ago
  46. 5d436ac Removed Die mock from MockAudioEncoder by Sebastian Jansson · 7 years ago
  47. 5283022 Shorten Chromium compile trybot names by Oleh Prypin · 7 years ago
  48. 39f491e Moved and simplifed the AEC3 API call skew estimator and added tests by Per Åhgren · 7 years ago
  49. 352314a Revert "VCMGenericDecoder threading updates for all but Android." by Lu Liu · 7 years ago
  50. 54daa3a Revert "Comment out DCHECK in dtor of VCMDecodedFrameCallback." by Lu Liu · 7 years ago
  51. c4f9824 Revert "Reduce locking in VideoReceiver and check the threading model." by Lu Liu · 7 years ago
  52. cf6e24a Forward the SignalNetworkRouteChanged from DtlsSrtpTransport to BaseChannel. by Zhi Huang · 7 years ago
  53. 52d8677 Fire OnRenegotiationNeeded when changing transceiver direction by Steve Anton · 7 years ago
  54. a1630f8 Reland "Base pacer padding in pause state on time since last send." by Sebastian Jansson · 7 years ago
  55. 3ab308f Inform the AEC3 echo remover about the status of the estimated delay by Per Åhgren · 7 years ago
  56. bbfccfd Added unittest to the AEC3 BlockProcessor class that tests longer calls by Per Åhgren · 7 years ago
  57. d5a272f Create public EncodedFrame interface. by philipel · 7 years ago
  58. 257cb10 Roll chromium_revision 04f0f4c72d..2c98648a24 (538005:538114) by Autoroller · 7 years ago
  59. c75f1e4 Reduce locking in VideoReceiver and check the threading model. by Tommi · 7 years ago
  60. d397a0d Add dropped frames metric on the receive side by Ilya Nikolaevskiy · 7 years ago
  61. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  62. 91bb667 Moved routes tracking to rtp transport controller. by Sebastian Jansson · 7 years ago
  63. 416332b Removed wait from congestion window test. by Sebastian Jansson · 7 years ago
  64. 5641fbb Add support for saving local audio input to file in AppRTCMobile by henrika · 7 years ago
  65. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  66. a425184 Fix override warnings. by Patrik Höglund · 7 years ago
  67. e5447fb Removed fake rtp transport controller send. by Sebastian Jansson · 7 years ago
  68. df023aa Extracted bitrate configuration from call class. by Sebastian Jansson · 7 years ago
  69. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  70. 9f016a0 Comment out DCHECK in dtor of VCMDecodedFrameCallback. by Tommi · 7 years ago
  71. a1f6661 Check that channel is in "send" before OKing DTMF by Harald Alvestrand · 7 years ago
  72. f906378 Split VCMCodecDataBase into VCMEncoderDataBase and VCMDecoderDataBase. by Niels Möller · 7 years ago
  73. a4e71b9 VCMGenericDecoder threading updates for all but Android. by Tommi · 7 years ago
  74. 0611065 Update JavaI420Buffer.allocate to use native allocations. by Sami Kalliomäki · 7 years ago
  75. defad84 Add batch script for running multiple VideoProcessor tests in parallel. by Rasmus Brandt · 7 years ago
  76. 2aa5666 Roll chromium_revision cff6369b11..04f0f4c72d (537896:538005) by Autoroller · 7 years ago
  77. 8ee1e5e Enable GetRemoteAudioSSLCertificate tests for Unified Plan by Steve Anton · 7 years ago
  78. 6e22137 Enable Unified Plan tests that were blocked on the stats collector by Steve Anton · 7 years ago
  79. 72a43a1 Collect packet loss and RTT stats of STUN binding requests. by Qingsi Wang · 7 years ago
  80. 54b8407 Clear current_direction when the RtpTransceiver is stopped by Steve Anton · 7 years ago
  81. 2f0d702 Parameterize PeerConnection integration tests for Unified Plan by Seth Hampson · 7 years ago
  82. afb0bb7 Remove PeerConnection voice_channel/video_channel methods by Steve Anton · 7 years ago
  83. db53f8e Add configurable STUN binding request interval. by Qingsi Wang · 7 years ago
  84. ed1ecea Roll chromium_revision 1f1e714a1e..cff6369b11 (537795:537896) by Autoroller · 7 years ago
  85. b6b00dc Safe behavior of the initial echo removal in AEC3 by Per Åhgren · 7 years ago
  86. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  87. c4bffed Roll chromium_revision fd6d802597..1f1e714a1e (537681:537795) by Autoroller · 7 years ago
  88. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  89. a05ee82 Fixed Digital mode of AGC2 implementation finished. by Alex Loiko · 7 years ago
  90. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  91. 6bd3cdd Remove special MD5 / SHA-1 digest classes. by Joachim Bauch · 7 years ago
  92. 0dd1b0a Revert "Revert "Enables PeerConnectionFactory using external fec controller"" by Ying Wang · 7 years ago
  93. 439f0bc Preparing for task queue in congenstion controller by Sebastian Jansson · 7 years ago
  94. 645898a Reduce severity of BWE start bitrate log to INFO. by Stefan Holmer · 7 years ago
  95. 694a36f Only log once per UpdateHistogram call. by Jonas Olsson · 7 years ago
  96. 52e5852 Adjust DTMF min inter-tone gap to 30 ms by Harald Alvestrand · 7 years ago
  97. b824b55 Delete unused sample project code. by Kári Tristan Helgason · 7 years ago
  98. 06a8f30 Moved analysis to Stats. by Sergey Silkin · 7 years ago
  99. 45d725d Support sending flexfec and simulcast together. by Danil Chapovalov · 7 years ago
  100. e127387 Added nisse@webrtc.org as owner in call. by Sebastian Jansson · 7 years ago