marpan@webrtc.org 993494764d Added metrics test code for the FEC packet masks. ...

The test computes metrics (average residual loss) for each mask type and size, 
for a given set of loss models (random and bursty), and verifies various 
behaviour of the codes (including relation/comparison to RS code).
Review URL: https://webrtc-codereview.appspot.com/748008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3185 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-11-28 19:43:58 +00:00

..

audio_coding

Opus mono/stereo on the same payloadtype, and fix of memory bug

2012-11-28 12:23:29 +00:00

audio_conference_mixer

Move src/ -> webrtc/

2012-10-22 18:19:23 +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

Added metrics test code for the FEC packet masks.

2012-11-28 19:43:58 +00:00

udp_transport

Add warning comment

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

utility

Remove operator overloading from RTPFragmentationHeader.

2012-11-23 19:17:23 +00:00

video_capture

Ensures that we can build using VS 2012 on Windows.

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

video_coding

Fixes two bugs related to padding in the jitter buffer.

2012-11-28 19:36:20 +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

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

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