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mflodman@webrtc.org d7d46887a6 Update receive only channels with RTT. 
vie_autotest_loopback.cc will not be part of the commit, it's only to show the test.

TEST=temporary with attached loopback test.

Review URL: https://webrtc-codereview.appspot.com/390007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1678 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 12:49:59 +00:00
..
build Removed warnings on Windows and enabled warnings-as-errors on Windows. 2012-02-07 14:56:45 +00:00
common_audio Optimized function WebRtcSpl_DownsampleFast for ARM-NEON platform. 2012-02-07 18:03:11 +00:00
common_video Added return due to gcc complaints in r1604. 2012-02-06 11:06:01 +00:00
modules Update receive only channels with RTT. 2012-02-14 12:49:59 +00:00
system_wrappers The pthread_t is non-pointer type. 2012-02-08 20:36:23 +00:00
video_engine Fixed incorrect packet loss reported to encoder. 2012-02-10 12:41:57 +00:00
voice_engine New attempt. 2012-02-10 15:21:33 +00:00
common_settings.gypi git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
common_types.h Removing unused code. 2012-01-17 12:45:47 +00:00
engine_configurations.h remove vie file API to take away media_file and utility modules. 2012-02-08 10:38:12 +00:00
LICENSE Aligning license file with file header 2011-11-02 09:31:39 +00:00
LICENSE_THIRD_PARTY Introduced ARM version of WebRtcSpl_SqrtFloor(). Function cycles reduced by ~ 30% in a real time VOE test in an android device (Nexus-S, ARMv7a). 2012-02-07 17:15:15 +00:00
PATENTS Adding copies of license files to src/ so that Chromium will get those as well. 2011-07-14 08:00:33 +00:00
README.chromium git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
typedefs.h git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-10-20 12:30:35 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.