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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / refs/heads/fp2-sibon-2.0.1 / common_types.h
  1. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  2. 9aa3497 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 10 years ago
  3. 2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  4. 7e68693 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  5. 93ae821 Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 by henrike@webrtc.org · 10 years ago
  6. 692224a Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  7. 9402619 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 10 years ago
  8. 2a0cbfc Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
  9. 3d6910c Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 10 years ago
  10. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  11. 18c2945 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  12. e2a7a77 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  13. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  14. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  15. c8ab721 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago
  16. 84350a9 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 10 years ago
  17. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 10 years ago
  18. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  19. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  20. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  21. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  22. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  23. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  24. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  25. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  26. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  27. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  28. b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  29. 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  30. 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  31. 21dc10d Make interface destructor virtual by sprang@webrtc.org · 11 years ago
  32. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  33. 2714c79 Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
  34. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  35. 5dcb4e3 webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 11 years ago
  36. ee6f8a2 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 11 years ago
  37. 52b2ee5 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
  38. 2d6f0df Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
  39. e422d12 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
  40. f292306 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 11 years ago
  41. fa2dd22 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
  42. e1198e6 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
  43. e3339fc Generic video-codec support. by pbos@webrtc.org · 11 years ago
  44. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  45. 8665399 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 11 years ago
  46. f4d3788 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 11 years ago
  47. ca0e88a VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 11 years ago
  48. 0049a76 Add number of inserted samples to NetEq statistics. by roosa@google.com · 12 years ago
  49. 90d333e Expose NetEq playout mode off through VoiceEngine. by roosa@google.com · 12 years ago
  50. bc687c5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago
  51. d75680a Clean up TraceCallback::Print. by andrew@webrtc.org · 12 years ago
  52. d898c01 Add libjingle-style stream-style logging. by andrew@webrtc.org · 12 years ago
  53. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago