turaj@webrtc.org 226db898f7 Dual-stream implementation, not including VoE APIs. ...

Review URL: https://webrtc-codereview.appspot.com/933015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3230 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-12-03 22:13:31 +00:00

..

audio_coding

Dual-stream implementation, not including VoE APIs.

2012-12-03 22:13:31 +00:00

audio_conference_mixer

Removing redundant codec unittest targets.

2012-12-03 10:52:29 +00:00

audio_device

Ensures that we can build using VS 2012 on Windows.

2012-11-26 09:12:02 +00:00

audio_processing

Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots.

2012-11-26 23:57:38 +00:00

bitrate_controller

Clarifies the bandwidth estimation interfaces.

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

interface

Remove operator overloading from RTPFragmentationHeader.

2012-11-23 19:17:23 +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

Add test to verify that padding only frames are passing through the RTP module.

2012-12-03 14:01:46 +00:00

udp_transport

Add warning comment

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

utility

Removing redundant codec unittest targets.

2012-12-03 10:52:29 +00:00

video_capture

Ensures that we can build using VS 2012 on Windows.

2012-11-26 09:12:02 +00:00

video_coding

Revert 3227

2012-12-03 20:08:57 +00:00

video_processing/main

Verify output frame timestamp in VideoProcessingModuleTest.Resampler.

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

video_render

Condition for DirectX variable on Windows

2012-11-23 09:41:42 +00:00

modules.gyp

Add test to verify that padding only frames are passing through the RTP module.

2012-12-03 14:01:46 +00:00