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
2015-11-04	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
2015-08-24	Update a ton of audio code to use size_t more correctly and in general reduce	Peter Kasting
2014-10-27	Add a simple AudioConverter class.	andrew@webrtc.org
2014-06-05	Fix the chain that propagates the audio frame's rtp and ntp timestamp including:	wu@webrtc.org
2014-05-30	Add a Reset() method to AudioFrame.	andrew@webrtc.org
2014-04-19	Resampler modifications in preparation for arbitrary audioproc rates.	andrew@webrtc.org
2014-04-03	Consolidate audio conversion from Channel and TransmitMixer.	andrew@webrtc.org
2013-07-03	Proper spacing for end-of-namespace comments.	pbos@webrtc.org
2013-05-21	Include files from webrtc/.. paths in voice_engine/	pbos@webrtc.org
2013-04-09	WebRtc_Word32 -> int32_t in voice_engine/	pbos@webrtc.org
2012-10-22	Move src/ -> webrtc/	andrew@webrtc.org