external/webrtc.git -

Age	Commit message (Expand)	Author
2016-01-13	Convert channel counts to size_t.	Peter Kasting
2015-11-27	Use webrtc/base/logging.h for voice_engine.	pbos
2015-11-24	Work around data race in TransmitMixer.	solenberg
2015-11-04	modules: more interface -> include renames	Henrik Kjellander
2015-10-28	system_wrappers: rename interface -> include	Henrik Kjellander
2015-09-23	Fix the maximum native sample rate in AudioProcessing	Alejandro Luebs
2015-09-23	Reduce LS_INFO spam from voice_engine/.	Peter Boström
2015-08-24	Update a ton of audio code to use size_t more correctly and in general reduce	Peter Kasting
2015-02-27	ProcessThread improvements.	tommi@webrtc.org
2015-02-26	Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away	kwiberg@webrtc.org
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