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webrtc / src.git / 40889f35fcd36979fdd466be9ea36beb63a3d5a5 / call / degraded_call.cc
  1. 836fee1 Calculate next process time in simulated network. by Sebastian Jansson · 6 years ago
  2. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  3. cc8e8bb Pass the media transport from JsepTransportController to Call. by Piotr (Peter) Slatala · 6 years ago
  4. 0378997 Adds flags indicating presence in allocation and feedback per packet. by Sebastian Jansson · 6 years ago
  5. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  6. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  7. 156d11d Adds packet_size to rtc::SentPacket in testing code. by Sebastian Jansson · 6 years ago
  8. 3229d65 Switch webrtc users from deprecated ctors. by Artem Titov · 7 years ago
  9. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 7 years ago
  10. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  11. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  12. eef09fc Fix race in DegradedCall::DestroyVideoSendStream by Erik Språng · 7 years ago
  13. 0970851 Reland: Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
  14. 16cba5c Revert "Add ability to emulate degraded network in Call via field trial" by Ilya Nikolaevskiy · 7 years ago
  15. 31a12c5 Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago