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stefan@webrtc.org 66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
..
build
Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout.
2013-07-03 21:50:33 +00:00
common_audio
Proper spacing for end-of-namespace comments.
2013-07-03 15:12:26 +00:00
common_video
Proper spacing for end-of-namespace comments.
2013-07-03 15:12:26 +00:00
modules
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
2013-07-05 14:30:48 +00:00
system_wrappers
Proper spacing for end-of-namespace comments.
2013-07-03 15:12:26 +00:00
test
Revert 4298 "Makes it possible to find files used by some unit t..."
2013-07-05 08:49:09 +00:00
tools
Revert 4298 "Makes it possible to find files used by some unit t..."
2013-07-05 08:49:09 +00:00
video_engine
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
2013-07-05 14:30:48 +00:00
voice_engine
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
2013-07-05 14:30:48 +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
Fix unnamed-type-template-args warnings on clang.
2013-06-27 10:18:09 +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 unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
2013-06-19 17:03:47 +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.