phoglund@webrtc.org 92bb417cb1 Decoupled RTP audio processor from RTP receiver. ...

BUG=
TEST=Ran vie_auto_test, rtp_rtcp_unittests, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3279 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-12-13 10:48:24 +00:00

..

audio_coding

ISAC_main_inst initialized to NULL to avoid potentially garbage pointer passed to WebRtcIsacfix_EncoderInit

2012-12-13 07:47:54 +00:00

audio_conference_mixer

Properly zero out unmixed frames.

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

audio_device

Not to enum NOTPRESENT audio devices with CoreAudio on Win

2012-12-07 22:36:07 +00:00

audio_processing

Added buildbot benchmarking in iSAC and APM into Android platform build.

2012-12-07 15:26:28 +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

Decoupled RTP audio processor from RTP receiver.

2012-12-13 10:48:24 +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

Vp8 tests: Removing legacy unused tests and reorganization of existing ones.

2012-12-13 00:37:22 +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