webrtc/src
pwestin@webrtc.org 0644b1dc35 Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.

An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).

Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.

The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):

Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo

Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-01 15:42:31 +00:00
..
build Use clang by default on Mac. 2011-11-30 01:16:06 +00:00
common_audio Made tables static. 2011-12-01 15:40:50 +00:00
common_video libyuv: Updating paths for test files 2011-11-29 17:50:07 +00:00
modules Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions 2011-12-01 15:42:31 +00:00
system_wrappers Fixes Valgrind warnings in system wrappers unittest. 2011-11-30 22:46:59 +00:00
video_engine Refactored ViESyncModule. 2011-11-30 18:31:36 +00:00
voice_engine Remove global voe::Channel::numSocketThreads. 2011-11-30 18:11:23 +00:00
common_settings.gypi git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
common_types.h Removing statistics API from NetEQ 2011-11-23 09:36:23 +00:00
engine_configurations.h To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory. 2011-11-30 15:35:44 +00:00
LICENSE Aligning license file with file header 2011-11-02 09:31:39 +00:00
LICENSE_THIRD_PARTY Add references to src/ copies for LICENSE etc. 2011-10-26 01:05:07 +00:00
PATENTS Adding copies of license files to src/ so that Chromium will get those as well. 2011-07-14 08:00:33 +00:00
README.chromium git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
typedefs.h git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-10-20 12:30:35 +00:00

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.