1. c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 4 years, 6 months ago
  2. 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 4 years, 6 months ago
  3. 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 6 months ago
  4. c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 6 months ago
  5. 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 4 years, 6 months ago
  6. f298855 Cleanup of feedback observer interface by Sebastian Jansson · 4 years, 9 months ago
  7. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 4 years, 10 months ago
  8. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 4 years, 10 months ago
  9. 425d6aa Add RtpPacketPacer interface for pacer control by Erik Språng · 5 years ago
  10. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
  11. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 5 years ago
  12. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  13. 59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 5 years ago
  14. 607a6f1 Moves conversion to ReceivedPacket from RtpPacketReceived to Call. by Sebastian Jansson · 5 years ago
  15. 4ad51d8 Removes SendSideCongestionController. by Sebastian Jansson · 5 years ago
  16. 8b27910 Include downlink delay into congestion window size. by Ying Wang · 5 years ago
  17. e896490 Revert "Fix target bitrate RTCP messages behavior for SVC streams" by Oleh Prypin · 5 years ago
  18. ab65d8a Fix target bitrate RTCP messages behavior for SVC streams by Ilya Nikolaevskiy · 5 years ago
  19. 2997ec9 Removes unused keep-alive from RtpTransportControllerSend. by Sebastian Jansson · 5 years ago
  20. 418dd0b Stop using special RTT value for DelayBasedBwe. by Sebastian Jansson · 5 years ago
  21. 487c09b Adds FakeNetworkPipeTest to rtc_unittests. by Sebastian Jansson · 5 years ago
  22. 836fee1 Calculate next process time in simulated network. by Sebastian Jansson · 5 years ago
  23. 813c79b Fix network emulation behavior when changing bandwidth. by Christoffer Rodbro · 5 years ago
  24. 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
  25. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  26. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  27. 0fc2843 Removing redundant argument for SSRCs from ctor of RtpVideoSender. by Amit Hilbuch · 6 years ago
  28. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  29. 44a262a Declares BitrateAllocator methods const. by Sebastian Jansson · 6 years ago
  30. 192eeec Enable End-to-End Encrypted Video Frames. by Benjamin Wright · 6 years ago
  31. 1298541 Removing unnecessary dependencies on socket.h. by Sebastian Jansson · 6 years ago
  32. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  33. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  34. c7ea852 Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe by Artem Titov · 6 years ago
  35. 3229d65 Switch webrtc users from deprecated ctors. by Artem Titov · 6 years ago
  36. b005087 Add replacements for all FakeNetworkPipe ctors. by Artem Titov · 6 years ago
  37. e23b8a9 Do not use FakeNetworkPipe::SetConfig. by Artem Titov · 6 years ago
  38. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  39. 9416ef8 Rename PayloadRouter to RtpVideoSender. by Stefan Holmer · 6 years ago
  40. 5ed25af Properly clean up RtpVideoSender. by Stefan Holmer · 6 years ago
  41. dbdb3a0 Refactoring PayloadRouter. by Stefan Holmer · 6 years ago
  42. bcf9180 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. by Alex Narest · 6 years ago
  43. b6b29e0 Convert video quality test from a TEST_F to a TEST fixture. by Patrik Höglund · 6 years ago
  44. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  45. 0c4f7be New api struct BitrateSettings. by Niels Möller · 6 years ago
  46. e625605 Moving task queue from Call to transport controller. by Sebastian Jansson · 6 years ago
  47. 0940811 Moving demux from FakeNetworkPipe to DirectTransport. by Sebastian Jansson · 6 years ago
  48. 8326780 Adds mock bitrate allocator. by Sebastian Jansson · 6 years ago
  49. 12130bb Reporting feedback availability to congestion controller. by Sebastian Jansson · 6 years ago
  50. 8d8cb56 Delete obsolete methods from MockRtpTransportControllerSend by Sebastian Jansson · 6 years ago
  51. 0970851 Reland: Add ability to emulate degraded network in Call via field trial by Erik Språng · 6 years ago
  52. 16cba5c Revert "Add ability to emulate degraded network in Call via field trial" by Ilya Nikolaevskiy · 6 years ago
  53. 31a12c5 Add ability to emulate degraded network in Call via field trial by Erik Språng · 6 years ago
  54. 19704ec Removing AvailableBandwidth method on transport controller. by Sebastian Jansson · 6 years ago
  55. 45087cd Moved retransmission rate limiter to Call class. by Sebastian Jansson · 6 years ago
  56. 832b1c8 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 2. by philipel · 6 years ago
  57. 0f9d9a9 Removed unused DeRegisterNetworkObserver. by Sebastian Jansson · 6 years ago
  58. 35dd6cd Added dependencies to mock transport controller send. by Sebastian Jansson · 6 years ago
  59. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 6 years ago
  60. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  61. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  62. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago