stefan@webrtc.org f72881406f Refactoring the receive-side bandwidth estimation into its own module. ... Each REMB group has one remote_bitrate_estimator object. For now the estimator keeps one estimate for every SSRC. In a later commit this will be unified and one estimate will be used for all SSRC in one group. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/605007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2363 4adac7df-926f-26a2-2b94-8c16560cd09d		2012-06-05 10:44:00 +00:00
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audio_coding	Minor style changes	2012-06-05 08:09:23 +00:00
audio_conference_mixer	Move audio_frame_operations to the utility module.	2012-05-29 22:13:14 +00:00
audio_device	Added gyp variable to include/exclude all tests.	2012-05-24 13:23:35 +00:00
audio_processing	Ignore return value of fwrites.	2012-06-01 02:41:14 +00:00
bitrate_controller	Attempt to fix broken encoding.	2012-06-04 11:04:05 +00:00
interface	Re-added ChangeUniqueId temporary for chrome.	2012-05-24 09:52:19 +00:00
media_file	Added gyp variable to include/exclude all tests.	2012-05-24 13:23:35 +00:00
remote_bitrate_estimator	Refactoring the receive-side bandwidth estimation into its own module.	2012-06-05 10:44:00 +00:00
rtp_rtcp	Refactoring the receive-side bandwidth estimation into its own module.	2012-06-05 10:44:00 +00:00
udp_transport	Fix compilation errors on ChromeOS	2012-05-30 16:46:09 +00:00
utility	Fixing gyp bug in https://webrtc-codereview.appspot.com/599006	2012-05-30 14:32:42 +00:00
video_capture	Added gyp variable to include/exclude all tests.	2012-05-24 13:23:35 +00:00
video_coding	Upgrade libvpx to cab6ac16 (v. 1.1.1 pre-point-release).	2012-06-01 07:43:02 +00:00
video_processing/main	Check return value of fwrite. [Video Module]	2012-05-29 17:33:13 +00:00
video_render	Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481.	2012-05-30 10:45:18 +00:00
modules.gyp	Refactoring the receive-side bandwidth estimation into its own module.	2012-06-05 10:44:00 +00:00