stefan@webrtc.org 84cd8e39cf Disable frame dropper for screenshare mode. ...

BUG=1466

Review URL: https://webrtc-codereview.appspot.com/1170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-03-07 13:12:32 +00:00

..

audio_coding

Changing non-const reference arguments to pointers, ACM

2013-02-21 10:27:48 +00:00

audio_conference_mixer

Replace AudioFrame's operator= with CopyFrom().

2013-01-22 04:44:30 +00:00

audio_device

Expose the capture-side AudioProcessing object and allow it to be injected.

2013-03-05 01:12:49 +00:00

audio_processing

Fix debug file buffer bug introduced in r3574.

2013-03-07 00:45:50 +00:00

bitrate_controller

Fix webrtc compilation errors for Chrome Win64

2013-01-29 06:45:22 +00:00

interface

Rename webrtc::StatsObserver to webrtc::CallStatsObserver

2013-02-19 22:09:36 +00:00

media_file

Fix webrtc compilation errors for Chrome Win64

2013-01-29 06:45:22 +00:00

pacing

Remove '<(library)' in gyp files.

2013-01-18 23:42:21 +00:00

remote_bitrate_estimator

Rename webrtc::StatsObserver to webrtc::CallStatsObserver

2013-02-19 22:09:36 +00:00

rtp_rtcp

Fix for build error on android introduced with r3609.

2013-03-05 09:30:47 +00:00

udp_transport

Bug fix for webrtc issue 1391. Typo in sin_length for socket address.

2013-02-15 18:42:12 +00:00

utility

Changing non-const reference arguments to pointers, ACM

2013-02-21 10:27:48 +00:00

video_capture

VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView.

2013-03-03 17:28:03 +00:00

video_coding

Disable frame dropper for screenshare mode.

2013-03-07 13:12:32 +00:00

video_processing/main

Remove '<(library)' in gyp files.

2013-01-18 23:42:21 +00:00

video_render

Misc cleanups to webrtc/android code:

2013-02-20 23:13:46 +00:00

modules.gyp

Initial upload of NetEq4

2013-01-29 12:09:21 +00:00