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gerrit-public.fairphone.software / platform / external / webrtc / 982639c0194f8f8507edbbda28493c890f8b5702 / call / test
  1. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  2. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  3. 0fc2843 Removing redundant argument for SSRCs from ctor of RtpVideoSender. by Amit Hilbuch · 6 years ago
  4. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  5. 44a262a Declares BitrateAllocator methods const. by Sebastian Jansson · 6 years ago
  6. 192eeec Enable End-to-End Encrypted Video Frames. by Benjamin Wright · 6 years ago
  7. 1298541 Removing unnecessary dependencies on socket.h. by Sebastian Jansson · 6 years ago
  8. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  9. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  10. c7ea852 Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe by Artem Titov · 6 years ago
  11. 3229d65 Switch webrtc users from deprecated ctors. by Artem Titov · 6 years ago
  12. b005087 Add replacements for all FakeNetworkPipe ctors. by Artem Titov · 6 years ago
  13. e23b8a9 Do not use FakeNetworkPipe::SetConfig. by Artem Titov · 6 years ago
  14. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  15. 9416ef8 Rename PayloadRouter to RtpVideoSender. by Stefan Holmer · 6 years ago
  16. 5ed25af Properly clean up RtpVideoSender. by Stefan Holmer · 6 years ago
  17. dbdb3a0 Refactoring PayloadRouter. by Stefan Holmer · 6 years ago
  18. 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
  19. b6b29e0 Convert video quality test from a TEST_F to a TEST fixture. by Patrik Höglund · 6 years ago
  20. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  21. 0c4f7be New api struct BitrateSettings. by Niels Möller · 7 years ago
  22. e625605 Moving task queue from Call to transport controller. by Sebastian Jansson · 7 years ago
  23. 0940811 Moving demux from FakeNetworkPipe to DirectTransport. by Sebastian Jansson · 7 years ago
  24. 8326780 Adds mock bitrate allocator. by Sebastian Jansson · 7 years ago
  25. 12130bb Reporting feedback availability to congestion controller. by Sebastian Jansson · 7 years ago
  26. 8d8cb56 Delete obsolete methods from MockRtpTransportControllerSend by Sebastian Jansson · 7 years ago
  27. 0970851 Reland: Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  28. 16cba5c Revert "Add ability to emulate degraded network in Call via field trial" by Ilya Nikolaevskiy · 7 years ago
  29. 31a12c5 Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  30. 19704ec Removing AvailableBandwidth method on transport controller. by Sebastian Jansson · 7 years ago
  31. 45087cd Moved retransmission rate limiter to Call class. by Sebastian Jansson · 7 years ago
  32. 832b1c8 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 2. by philipel · 7 years ago
  33. 0f9d9a9 Removed unused DeRegisterNetworkObserver. by Sebastian Jansson · 7 years ago
  34. 35dd6cd Added dependencies to mock transport controller send. by Sebastian Jansson · 7 years ago
  35. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  36. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  37. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  38. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago