andrew@webrtc.org aada86b261 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
2014-10-27 18:18:17 +00:00
2014-10-27 17:22:15 +00:00
2014-10-27 18:18:17 +00:00
2014-09-23 05:56:44 +00:00
2014-06-17 08:54:03 +00:00
2014-08-25 14:41:41 +00:00
2014-10-01 08:03:19 +00:00
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
BSD-3-Clause 446 MiB
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