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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / d084db95dcfb5f96b5deab5a91a5218404f2ca21 / voice_engine / include
  1. 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
  2. 66ccaff Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  3. 4ff0eda Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  4. 2e4c621 (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
  5. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  6. 9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
  7. 4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
  8. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  9. 87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 11 years ago
  10. 942ba53 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 11 years ago
  11. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  12. 7b72264 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  13. b43ac9f Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  14. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  15. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  16. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  17. b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  18. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  19. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  20. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  21. 2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 years ago
  22. d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
  23. 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
  24. 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
  25. ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
  26. f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
  27. 54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
  28. e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
  29. 3b6f728 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  30. 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
  31. 2ffc8bf Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
  32. 365ca40 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
  33. 31b4448 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
  34. 13f66d1 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  35. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  36. 2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  37. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
  38. 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
  39. b79627b Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
  40. b9e5a3d Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 11 years ago
  41. ead8a5b Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. by turaj@webrtc.org · 11 years ago
  42. b9e3afc Add GetAudioFrame API to VoiceEngine. by roosa@google.com · 12 years ago
  43. ca77149 Add API to retreive last received RTP timestamp to VoiceEngine. by roosa@google.com · 12 years ago
  44. 7db5290 VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
  45. 15e35cc Expose Set and Get Recording/Playout sample rate apis by leozwang@webrtc.org · 12 years ago
  46. c65ae4b Revert 3231 - VoE Changes to enable dual_streaming. by perkj@webrtc.org · 12 years ago
  47. d6f028b VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
  48. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago