stefan@webrtc.org a4b58860b7 Add a counter to the video rtp play output filename. ...

Review URL: https://webrtc-codereview.appspot.com/1040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3379 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-01-17 09:27:17 +00:00

..

audio_coding

Adding AUDIO application as default for Opus stereo

2012-12-20 09:23:10 +00:00

audio_conference_mixer

Properly zero out unmixed frames.

2012-12-07 19:37:16 +00:00

audio_device

Clean up variable after it gets deleted

2012-12-19 05:36:36 +00:00

audio_processing

Fix simulated analog gain in audioproc.

2013-01-11 23:11:29 +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

Replace voice engine utility functions with system wrapper variants.

2013-01-02 16:06:39 +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

Reformatted rtp_rtcp_impl*.

2013-01-16 10:27:33 +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

Add a counter to the video rtp play output filename.

2013-01-17 09:27:17 +00:00

video_processing/main

Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers.

2013-01-08 19:19:59 +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