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webrtc_m130/webrtc/modules
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mikhal@webrtc.org adc64a7216 VCM/Timing: Setting clear names to members & methods 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1524004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-30 16:20:18 +00:00
..
audio_coding
Remove #pragma once
2013-05-29 13:19:09 +00:00
audio_conference_mixer
Include files from webrtc/.. paths in audio_conference_mixer/
2013-05-28 08:13:20 +00:00
audio_device
Remove #pragma once
2013-05-29 13:19:09 +00:00
audio_processing
Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD
2013-05-28 09:24:03 +00:00
bitrate_controller
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
desktop_capture
Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith().
2013-05-24 21:07:20 +00:00
interface
Breaking out RTP header parsing from the RTP module.
2013-05-29 12:12:51 +00:00
media_file
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
pacing
Fix crash in pacer.
2013-05-06 16:37:22 +00:00
remote_bitrate_estimator
Breaking out RTP header parsing from the RTP module.
2013-05-29 12:12:51 +00:00
rtp_rtcp
Include files from webrtc/.. paths in rtp_rtcp/
2013-05-29 14:27:38 +00:00
utility
Consolidate common_audio into a single target.
2013-04-30 23:43:26 +00:00
video_capture
Avoid NPE crash on Android platforms that don't support getting preview framerate.
2013-05-17 17:33:31 +00:00
video_coding
VCM/Timing: Setting clear names to members & methods
2013-05-30 16:20:18 +00:00
video_processing/main
Include files from webrtc/.. paths in video_processing/
2013-05-27 14:12:16 +00:00
video_render
Log timestamp of the frame when it's dropped from the render module
2013-05-21 00:16:01 +00:00
module_common_types_unittest.cc
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
2013-04-11 17:48:02 +00:00
modules.gyp
Add DesktopCapturer interface for desktop capturers.
2013-04-29 20:10:57 +00:00