webrtc

History

stefan@webrtc.org a5cb98cbbd Breaking out RTP header parsing from the RTP module.

This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-05-29 12:12:51 +00:00

build

Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.

2013-05-29 00:38:02 +00:00

common_audio

Include files from webrtc/.. paths in common_audio/

2013-05-27 09:49:58 +00:00

common_video

Recalibrate point sample expectation

2013-05-20 18:17:44 +00:00

modules

Breaking out RTP header parsing from the RTP module.

2013-05-29 12:12:51 +00:00

system_wrappers

Include files from webrtc/.. paths in system_wrappers/

2013-05-27 15:07:45 +00:00

test

Include files from webrtc/.. paths in test/channel_transport/

2013-05-27 15:02:23 +00:00

tools

Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.

2013-05-29 00:38:02 +00:00

video_engine

Breaking out RTP header parsing from the RTP module.

2013-05-29 12:12:51 +00:00

voice_engine

Breaking out RTP header parsing from the RTP module.

2013-05-29 12:12:51 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

Replace ExtraCodecOptions with new Config class that supports multiple settings at once.

2013-05-14 08:02:25 +00:00

common_unittest.cc

Added a config class to ease passing a set of options across webrtc.

2013-05-13 08:06:36 +00:00

common.h

Added a config class to ease passing a set of options across webrtc.

2013-05-13 08:06:36 +00:00

engine_configurations.h

Remove WEBRTC_*_ENGINE_NETWORK_API use

2013-04-04 19:51:42 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

PATENTS

Move src/ -> webrtc/

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

README.chromium

Move src/ -> webrtc/

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

typedefs.h

Disable clang C++11 warnings to permit OVERRIDE keyword.

2013-05-08 08:34:34 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.