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webrtc_m130/webrtc/modules
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pwestin@webrtc.org 52b4e8871a Adding trace and changing pacing constants 
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 19:02:17 +00:00
..
audio_coding
Consolidate common_audio into a single target.
2013-04-30 23:43:26 +00:00
audio_conference_mixer
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
audio_device
Consolidate common_audio into a single target.
2013-04-30 23:43:26 +00:00
audio_processing
Consolidate common_audio into a single target.
2013-04-30 23:43:26 +00:00
bitrate_controller
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
desktop_capture
Add DesktopCapturer interface for desktop capturers.
2013-04-29 20:10:57 +00:00
interface
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
2013-04-11 17:48:02 +00:00
media_file
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
pacing
Adding trace and changing pacing constants
2013-05-02 19:02:17 +00:00
remote_bitrate_estimator
Apply Chromium C++ style to RemoteRateControl.
2013-04-30 08:33:46 +00:00
rtp_rtcp
Adding trace and changing pacing constants
2013-05-02 19:02:17 +00:00
utility
Consolidate common_audio into a single target.
2013-04-30 23:43:26 +00:00
video_capture
Make sure timestamps are monotonically increasing.
2013-04-24 23:54:31 +00:00
video_coding
Revert 3933 "Remove traces of deprecated WebRtc_Word types."
2013-05-02 09:30:12 +00:00
video_processing/main
Consolidate common_audio into a single target.
2013-04-30 23:43:26 +00:00
video_render
WebRTCDemo: handle stride!=width from first frame.
2013-04-10 23:21:10 +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