Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / b66003ca79cd34f65ef964a5e3b4766bc97a5659 / call
  1. 1a1c52b H.264 temporal layers w/frame marking (PART 2/3) by Johnny Lee · 6 years ago
  2. 836fee1 Calculate next process time in simulated network. by Sebastian Jansson · 6 years ago
  3. 7ff164e Plumbing of feedback on request setting by Johannes Kron · 6 years ago
  4. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  5. b769894 Remove rule that discourages passing optional by const reference by Danil Chapovalov · 6 years ago
  6. 681de20 Stop changing the requested max bitrate based on protection level. by Rasmus Brandt · 6 years ago
  7. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 6 years ago
  8. 05cf6be [clang-tidy] Apply performance-move-const-arg fixes. by Mirko Bonadei · 6 years ago
  9. 6347029 Removes usages of TaskQueueCongestionControl field trial. by Sebastian Jansson · 6 years ago
  10. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
  11. eceea31 Reduces locking in SimulatedNetwork class. by Sebastian Jansson · 6 years ago
  12. 813c79b Fix network emulation behavior when changing bandwidth. by Christoffer Rodbro · 6 years ago
  13. aa01f27 Removes all const Clock*. by Sebastian Jansson · 6 years ago
  14. 8c8feb9 Moves packet overhead from network nodes to simulation. by Sebastian Jansson · 6 years ago
  15. 949f0fd Move FrameCountObserver from RTPSender to RtpVideoSender by Niels Möller · 6 years ago
  16. f5b216a Pass explicit frame dependency information to RtpPayloadParams by Elad Alon · 6 years ago
  17. 48c5493 Add 'UpdateAllocationLimits' in media transport. by Piotr (Peter) Slatala · 6 years ago
  18. 739baf0 [clang-tidy] Apply performance-for-range-copy fixes. by Mirko Bonadei · 6 years ago
  19. f380284 (7) Rename files to snake_case: remove forwarding headers by Steve Anton · 6 years ago
  20. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  21. a8f9e25 Make sure lost packets are removed from FakeNetworkPipe. by Johannes Kron · 6 years ago
  22. 1e27fec Negate flag name for prerender smoothing and update comments. by Rasmus Brandt · 6 years ago
  23. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  24. 0500b52 Reduce webrtc_perf_tests duration on buildbots by Ilya Nikolaevskiy · 6 years ago
  25. 05acd2b Removes clock from TransportFeedbackAdapter. by Sebastian Jansson · 6 years ago
  26. d15687d Don't include packetization overhead in protection bitrate. by Erik Språng · 6 years ago
  27. ecb6897 Adds repeating task class. by Sebastian Jansson · 6 years ago
  28. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  29. 77536a2 Rename EncodedImage::_length --> size_, and make private. by Niels Möller · 6 years ago
  30. 921d366 Remove comments about using std::shared_ptr. by Mirko Bonadei · 6 years ago
  31. aec15aa (5) Rename files to snake_case: install forwarding headers by Steve Anton · 6 years ago
  32. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  33. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago
  34. 4687915 Enable use of MediaTransportInterface for video streams. by Niels Möller · 6 years ago
  35. 53eae87 Add PeerConnection option to enable RTX handling in the audio jitter buffer. by Jakob Ivarsson · 6 years ago
  36. c610e26 Include pacing buffer size in congestion window. by Christoffer Rodbro · 6 years ago
  37. c12d41b Add field trial kill switch for packetization overhead subtraction. by Erik Språng · 6 years ago
  38. f331de6 Remove unused VideoReceiveStream::Config::AddRtxBinding. by Rasmus Brandt · 6 years ago
  39. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  40. 482b3ef Account for packetization overhead when setting target bitrate. by Erik Språng · 6 years ago
  41. 412d185 Delete pre_encode_callback from VideoSendStream::Config by Niels Möller · 6 years ago
  42. 29e13fd Delete rtc::PacketTime (was an alias for int64_t) by Niels Möller · 6 years ago
  43. 0fc2843 Removing redundant argument for SSRCs from ctor of RtpVideoSender. by Amit Hilbuch · 6 years ago
  44. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  45. 02c4f15 Stop using deprecated PacedSender method from RtpTransportControllerSend. by Sebastian Jansson · 6 years ago
  46. b275788 Register stat callbacks after rate observer is registered. by Piotr (Peter) Slatala · 6 years ago
  47. 3d2ed19 Remove Transport implementation from ChannelSend by Fredrik Solenberg · 6 years ago
  48. 8eeccbe Delete Start and Stop methods from TestVideoCapturer. by Niels Möller · 6 years ago
  49. 1618095 Cleanup of RtpTransportControllerSend. by Sebastian Jansson · 6 years ago
  50. 2701bc9 Signals start rate when registering to TargetTransferRateObserver. by Sebastian Jansson · 6 years ago
  51. 00672b1 Don't trigger too many probes when max allocated bitrate changes. by Erik Språng · 6 years ago
  52. 514f084 New statistic added to VideoReceiveStream to determine latency to first decode. by Benjamin Wright · 6 years ago
  53. d1d7b23 Include protection bitrate in total max allocated bitrate by Erik Språng · 6 years ago
  54. 87609be Merges RtpTransportControllerSend with SendSideCongestionController. by Sebastian Jansson · 6 years ago
  55. af2adda Explicit comparisons on NetworkRoute. by Sebastian Jansson · 6 years ago
  56. d0b69a8 Send and receive color space information if available by Johannes Kron · 6 years ago
  57. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  58. 10403ae Add PeerConnection option to configure minimum audio jitter buffer delay. by Jakob Ivarsson · 6 years ago
  59. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  60. 53382cb Move RtcpStatistics from common_types.h to a new header file by Niels Möller · 6 years ago
  61. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  62. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  63. af228ee Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN. by Alex Loiko · 6 years ago
  64. 8b5d9d8 Remove the audio/video split for the RTCP report intervals. by Jiawei Ou · 6 years ago
  65. 6736df1 Moves BitrateAllocationUpdate to api. by Sebastian Jansson · 6 years ago
  66. 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
  67. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  68. 89c94b9 Adds target bandwidth to BitrateAllocator. by Sebastian Jansson · 6 years ago
  69. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  70. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  71. cc8e8bb Pass the media transport from JsepTransportController to Call. by Piotr (Peter) Slatala · 6 years ago
  72. de8e6e6 Refactor bitrate configuration in CallTest by Niels Möller · 6 years ago
  73. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  74. c2ebe21 Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" by Jiawei Ou · 6 years ago
  75. cd2e105 Reenable test RampUpTest.AudioTransportSequenceNumber by Niels Möller · 6 years ago
  76. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  77. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  78. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  79. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  80. 2cd3b4c Fixing bug in SimulatedNetwork where packets stop. by Sebastian Jansson · 6 years ago
  81. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  82. 59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 6 years ago
  83. be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 6 years ago
  84. 7182286 Allow FakeNetworkPipe to wake up its processing thread by Sebastian Jansson · 6 years ago
  85. 69807e8 Depend directly on destination targets. by Yves Gerey · 6 years ago
  86. 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
  87. 992a868 Fix for clock reset repair. by Christoffer Rodbro · 6 years ago
  88. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  89. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  90. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  91. e2754c9 Fixes bug in AudioPriorityBitrateAllocationStrategy field trial. by Sebastian Jansson · 6 years ago
  92. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  93. 1803bb2 Fix for clock read race in FakeNetworkPipe. by Christoffer Rodbro · 6 years ago
  94. 3284b61 Fix for packet loss tracking in network emulation. by Christoffer Rodbro · 6 years ago
  95. 2506839 Add DCHECK for wrap around in RtpVideoSender::OnBitrateUpdated. by Bjorn Terelius · 6 years ago
  96. 44a262a Declares BitrateAllocator methods const. by Sebastian Jansson · 6 years ago
  97. 362cb50 Remove redundant RTC_DCHECK of max/min RTP header extension id by Johannes Kron · 6 years ago
  98. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  99. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  100. 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago