external/webrtc.git -

Age	Commit message (Expand)	Author
2016-01-13	Convert channel counts to size_t.	Peter Kasting
2015-12-16	Bugfix that fixes the error where the audio processing module is called	peah
2015-12-14	Correcting the check for the return code produced by	peah
2015-11-04	modules: more interface -> include renames	Henrik Kjellander
2015-10-28	system_wrappers: rename interface -> include	Henrik Kjellander
2015-08-31	Stylizing AudioConferenceMixer.	minyuel
2015-08-24	Update a ton of audio code to use size_t more correctly and in general reduce	Peter Kasting
2015-08-14	Integrate Intelligibility with APM	ekmeyerson
2015-07-28	Removing AudioMixerStatusReceiver and ParticipantStatistics.	Minyue
2014-05-14	Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation ...	henrika@webrtc.org
2014-04-22	Support arbitrary input/output rates and downmixing in AudioProcessing.	andrew@webrtc.org
2014-04-14	Added a new OnMoreData() interface which will not feed the playout data to APM.	xians@webrtc.org
2014-04-03	Consolidate audio conversion from Channel and TransmitMixer.	andrew@webrtc.org
2014-03-18	Resolves TSan v2 warnings in voe_auto_test.	henrika@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
2013-04-29	Add a wrapper around PushSincResampler and the old Resampler.	andrew@webrtc.org
2013-04-09	WebRtc_Word32 -> int32_t in voice_engine/	pbos@webrtc.org
2013-01-22	Replace AudioFrame's operator= with CopyFrom().	andrew@webrtc.org
2012-10-22	Move src/ -> webrtc/	andrew@webrtc.org