Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / test / fake_network_pipe.cc
  1. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  2. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  3. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/test/fake_network_pipe.cc]
  4. 19f5143 Keep track of the capacity delay error in the FakeNetworkPipe. by philipel · 7 years ago
  5. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  6. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  7. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  8. 20c84cc Making FakeNetworkPipe demux audio and video packets. by minyue · 7 years ago
  9. e5ad5ca Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 7 years ago
  10. 3a3bd50 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 7 years ago
  11. 9c47b00 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 7 years ago
  12. 50235b7 Make FakeNetworkPipe not busy loop any more. by philipel · 7 years ago
  13. 5ef2bc1 Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) by philipel · 7 years ago
  14. b80bdca Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) by philipel · 7 years ago
  15. a518a39 Fixes a bug where a video stream can get stuck in the suspended state. by stefan · 7 years ago
  16. e9ad271 Increase the send-time history to 60 seconds. by stefan · 7 years ago
  17. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  18. 90ce01d The current default schedule delay of 30 ms prohibits by isheriff · 8 years ago
  19. a6a7007 Fix FakeNetworkPipe to not deliver packet faster than requested. by danilchap · 8 years ago
  20. 536378b Allow FakeNetworkPipe to drop packets in bursts. by philipel · 8 years ago
  21. a2c5523 Allow packets to be reordered in the fake network pipe. by philipel · 8 years ago
  22. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  23. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  24. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  25. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 9 years ago
  26. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 9 years ago
  27. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 9 years ago
  28. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  29. bfe6e08 Add simulation of network effects to video_loopback tool. by stefan@webrtc.org · 10 years ago
  30. b8e9e44 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 10 years ago
  31. c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
  32. faada6e Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago