aada86b261146336d74ea09acedaf40e5c2f4618 - src

commit	aada86b261146336d74ea09acedaf40e5c2f4618	[log] [tgz]
author	andrew@webrtc.org <andrew@webrtc.org>	Mon Oct 27 18:18:17 2014
committer	andrew@webrtc.org <andrew@webrtc.org>	Mon Oct 27 18:18:17 2014
tree	add156d8ecf1be1623ad83f78e97656d2b08bf7e
parent	33a0e2d7efa9add39ec315c12d9ef06aa783f687 [diff]

Add a simple AudioConverter class.

This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.

The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.

BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/30779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_audio/BUILD.gn[diff]
webrtc/common_audio/audio_converter.cc[Added - diff]
webrtc/common_audio/audio_converter.h[Added - diff]
webrtc/common_audio/audio_converter_unittest.cc[Added - diff]
webrtc/common_audio/common_audio.gyp[diff]
webrtc/voice_engine/utility.cc[diff]

6 files changed