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gerrit-public.fairphone.software / platform / external / webrtc / 97899a09fbd8ae8b244db3062616fd497d305bbb / call / call.h
  1. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  2. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 7 years ago
  3. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  4. fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
  5. e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
  6. 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
  7. 0dd1b0a Revert "Revert "Enables PeerConnectionFactory using external fec controller"" by Ying Wang · 7 years ago
  8. 0073301 Revert "Enables PeerConnectionFactory using external fec controller" by Taylor Brandstetter · 7 years ago
  9. 4f07bdb Enables PeerConnectionFactory using external fec controller by Ying Wang · 7 years ago
  10. 8366e17 Rename Call::Config to CallConfig, keep old name as alias. by Niels Möller · 7 years ago
  11. 3b790f3 Make fec controller plug-able. by Ying Wang · 7 years ago
  12. 292a73e Deliver packet to Call as rtc::CopyOnWriteBuffer by Danil Chapovalov · 7 years ago
  13. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  14. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  15. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  16. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  17. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  18. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  19. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  20. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/call.h]
  21. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  22. db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  23. e5c4a81 Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 7 years ago
  24. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  25. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  26. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  27. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  28. a5e0df6 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 7 years ago
  29. 4b97980 Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 7 years ago
  30. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 7 years ago
  31. 9641c13 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 7 years ago
  32. 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
  33. b8a654c Delete declaration of non-existing function webrtc::Version(). by nisse · 8 years ago
  34. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 8 years ago
  35. 7250b39 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  36. 446fcb6 Clean up FlexfecReceiveStream ctor signatures. by brandtr · 8 years ago
  37. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago