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voice_engine
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transmit_mixer.cc
Age
Commit message (
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Author
2014-07-11
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
tommi@webrtc.org
2014-05-12
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation ...
henrika@webrtc.org
2014-04-30
Only clamp to 16 kHz when AECM is enabled.
andrew@webrtc.org
2014-04-25
Replace scoped_array<T> with scoped_ptr<T[]>.
andrew@webrtc.org
2014-04-17
Removes parts of the VoEBase sub API as part of a clean-up operation where th...
henrika@webrtc.org
2014-04-03
Consolidate audio conversion from Channel and TransmitMixer.
andrew@webrtc.org
2014-04-02
Extends max sample rate from 96kHz to 192kHz on the input side.
henrika@webrtc.org
2014-03-18
Resolves TSan v2 warnings in voe_auto_test.
henrika@webrtc.org
2014-02-11
Add an interface for accepting keypress signals to AudioProcessing.
andrew@webrtc.org
2014-01-30
Move out typing detection to its own class.
henrikg@webrtc.org
2014-01-11
Minor voice engine improvements around AGC.
andrew@webrtc.org
2014-01-07
Remove the requirement to call set_sample_rate_hz and friends.
andrew@webrtc.org
2013-10-04
Clean up AudioProcessing defaults and errors.
andrew@webrtc.org
2013-09-17
Adds a new voice engine warning for the typing noise off state.
jiayl@webrtc.org
2013-08-07
Ref-counted rewrite of ChannelManager.
pbos@webrtc.org
2013-07-31
Merge r4394 from stable to trunk.
xians@webrtc.org
2013-07-31
Merge r4326 from stable to trunk.
xians@webrtc.org
2013-07-03
Proper spacing for end-of-namespace comments.
pbos@webrtc.org
2013-05-14
Remove const for plain data types in voice_engine/
pbos@webrtc.org
2013-05-07
Refactoring for typing detection
niklas.enbom@webrtc.org
2013-05-02
Replace Resampler with PushResampler in transmit_mixer.
andrew@webrtc.org
2013-04-09
WebRtc_Word32 -> int32_t in voice_engine/
pbos@webrtc.org
2013-04-04
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
henrika@webrtc.org
2013-03-02
Don't upsample the capture signal early.
andrew@webrtc.org
2013-03-01
Properly error check calls to AudioProcessing.
andrew@webrtc.org
2013-01-22
Replace AudioFrame's operator= with CopyFrom().
andrew@webrtc.org
2013-01-09
Use %d for signed value in trace.
andrew@webrtc.org
2012-11-07
Move capture level computation after all processing.
andrew@webrtc.org
2012-10-22
Move src/ -> webrtc/
andrew@webrtc.org