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webrtc / src / a70695a3e1e7ead375d4bbc7997cd630d1ac1731 / webrtc / call.h
  1. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  2. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  3. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  4. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  5. 2e5cfcd Add periodic logging of video stats. by asapersson · 9 years ago
  6. 9c0b551 Fix bug where min transmit bitrate wasn't working by sprang · 9 years ago
  7. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 9 years ago
  8. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  9. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  10. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  11. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  12. 7a43d25 Make the audio channel communicate network state changes to the call. by skvlad · 9 years ago
  13. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  14. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  15. cf18b34 Align new VoE API with design. by solenberg · 9 years ago
  16. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 10 years ago
  17. e526974 Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/ by solenberg · 10 years ago
  18. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 10 years ago
  19. cd67022 Define Stream base classes by Jelena Marusic · 10 years ago
  20. 78fb3b3 C++11 in-class member initialization in Call configs. by Fredrik Solenberg · 10 years ago
  21. 04f4931 VoE2 API draft by Fredrik Solenberg · 10 years ago
  22. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  23. 76c53d3 Remove ViE interface usage from VideoReceiveStream. by Peter Boström · 10 years ago
  24. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  25. 16825b1 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 10 years ago
  26. 2b19f06 Wire up RTT statistics to webrtc::Call. by pbos@webrtc.org · 10 years ago
  27. 0087318 Implement settable min/start/max bitrates in Call. by pbos@webrtc.org · 10 years ago
  28. 0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  29. a73a678 Remove -1 from Call::Config::start_bitrate_bps. by pbos@webrtc.org · 10 years ago
  30. 42684be Wire up CPU adaptation in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  31. bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 11 years ago
  32. 26c0c41 Network up/down signaling in Call. by pbos@webrtc.org · 11 years ago
  33. bd249bc Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 11 years ago
  34. eb16b81 Implements start bitrate for new video API. by mflodman@webrtc.org · 11 years ago
  35. 6ae48c6 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 11 years ago
  36. caba2d2 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 11 years ago
  37. 5ca6a53 Remove TraceCallback use from Call. by pbos@webrtc.org · 11 years ago
  38. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  39. bdc5ed2 Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 11 years ago
  40. 39669c5 Remove outdated DestroyVideoSendStream comment. by pbos@webrtc.org · 11 years ago
  41. b429e51 cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  42. 7e9315b Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  43. 2c46f8d Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
  44. 5a63655 Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
  45. 6488761 Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
  46. 16e03b7 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago[Renamed (95%) from webrtc/video_engine/new_include/call.h]
  47. de74b64 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  48. 841c8a4 Rename VideoCall to Call. by pbos@webrtc.org · 12 years ago[Renamed (84%) from webrtc/video_engine/new_include/video_call.h]
  49. 74fa489 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 12 years ago
  50. fd39e13 Remove VideoEngine class from new VideoEngine API. by pbos@webrtc.org · 12 years ago[Renamed (75%) from webrtc/video_engine/new_include/video_engine.h]
  51. 4052370 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 12 years ago
  52. 6879c8a Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 12 years ago
  53. 025f4f1 Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 12 years ago
  54. 1ecee9a Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 12 years ago
  55. 29d5839 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 12 years ago
  56. 65f995a New ViE interface. by mflodman@webrtc.org · 12 years ago