mflodman@webrtc.org 1c61196095 Removed not used include. ...

TEST=Compiles.

Review URL: https://webrtc-codereview.appspot.com/966025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3150 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-11-22 09:37:27 +00:00

..

audio_coding

Ensure opus_demo has a targets block.

2012-11-21 20:16:53 +00:00

audio_conference_mixer

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

audio_device

Add Android OWNER files

2012-11-09 17:51:55 +00:00

audio_processing

Porting ARM optimization from Android to ios.

2012-11-17 00:22:46 +00:00

bitrate_controller

Clarifies the bandwidth estimation interfaces.

2012-11-13 15:02:13 +00:00

interface

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

media_file

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

pacing

Adding pacing module, will replace the transmission_bucket in the RTP module.

2012-11-09 20:56:23 +00:00

remote_bitrate_estimator

Move SSRC list to RemoteBitrateEstimator.

2012-11-19 10:09:20 +00:00

rtp_rtcp

Removed not used include.

2012-11-22 09:37:27 +00:00

udp_transport

Add warning comment

2012-11-16 22:46:36 +00:00

utility

Add Android include path so that header files can follow google style

2012-11-15 18:17:40 +00:00

video_capture

Setting capture stride to width

2012-11-21 22:18:32 +00:00

video_coding

Setting correct stride for VP8 encoder

2012-11-21 00:15:19 +00:00

video_processing/main

Verify output frame timestamp in VideoProcessingModuleTest.Resampler.

2012-11-08 01:07:21 +00:00

video_render

Fix OpenGL rendering of WebRTCDemo by accounting for stride != width.

2012-11-12 16:39:25 +00:00

modules.gyp

Adding pacing module, will replace the transmission_bucket in the RTP module.

2012-11-09 20:56:23 +00:00