Log - b7e5b2741f9cbd68e7be54e3445d28d266477c92 - fp2-dev/platform/external/chromium_org/third_party/webrtc

81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
d2632a0 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
467f756 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
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
91d88e1 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 10 years ago
f7c73b5 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
cceb392 Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 10 years ago
ad065d0 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 10 years ago
87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 years ago
942ba53 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. by xians@webrtc.org · 10 years ago
22470b5 Minor voice engine improvements around AGC. by andrew@webrtc.org · 11 years ago
c7d7363 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
b43ac9f Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
3d553d4 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
e06943f Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
0e6fa8c Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
44f1239 Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
28832e1 Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
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
0f919be Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 11 years ago
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
b79627b Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
b9e5a3d Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 11 years ago
b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago