mflodman@webrtc.org 84dc3d134d Add REMB functionality to ViE. ... This CL only adds support for encoding one stream, but receiving multiple streams. BUG= TEST=video_engine_core_unittest + auto_test/loopback Review URL: http://webrtc-codereview.appspot.com/333011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-12-22 10:26:13 +00:00
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audio_coding	Converted to gtest, writing output files properly and no longer uses exceptions.	2011-12-21 13:34:18 +00:00
audio_conference_mixer	Removed usage of the deprecated critical section constructor in audio_conference_mixer.	2011-12-16 23:00:30 +00:00
audio_device	Disable API tests on ALSA since the tests don't work for all the alsa devices.	2011-12-09 14:05:29 +00:00
audio_processing	Fix error in RtpDump::Start due to r1156.	2011-12-13 22:59:33 +00:00
interface	Review URL: http://webrtc-codereview.appspot.com/295010	2011-12-16 14:31:37 +00:00
media_file	Removed usage of the deprecated critical section constructor in media_file.	2011-12-14 17:27:58 +00:00
rtp_rtcp	Add REMB functionality to ViE.	2011-12-22 10:26:13 +00:00
udp_transport	Removed unused function messing up the symbols.	2011-12-22 09:48:48 +00:00
utility	Removed usage of the deprecated critical section constructor in modules/utility.	2011-12-16 19:53:46 +00:00
video_capture	Remove warnings in VideoEngine, capture module and render module.	2011-12-09 10:12:57 +00:00
video_coding	Remove unused variable.	2011-12-22 08:34:31 +00:00
video_processing/main	Including Brighten function in namespace VideoProcessing	2011-12-20 15:33:49 +00:00
video_render	Fix last Mac/clang compile error.	2011-12-20 22:23:21 +00:00
modules.gyp	Restructuring and adding dummy unit test target.	2011-11-17 13:56:54 +00:00