6 Commits

Author SHA1 Message Date
pwestin@webrtc.org
aafa5a331c Coverty report: Unititialized members
Review URL: http://webrtc-codereview.appspot.com/349007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1436 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 07:07:37 +00:00
asapersson@webrtc.org
5249cc8f77 Review URL: http://webrtc-codereview.appspot.com/295010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1219 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-16 14:31:37 +00:00
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
mflodman@webrtc.org
a02ef1ace2 Fix broken tree.
Review URL: http://webrtc-codereview.appspot.com/267015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@943 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 07:50:50 +00:00
mflodman@webrtc.org
1f69c03739 Added size sanity check for copying app specific RTCP data.
Similar check as done in RTCPUtility::RTCPParserV2::ParseAPPItem.

Review URL: http://webrtc-codereview.appspot.com/277002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@942 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 06:12:39 +00:00
niklase@google.com
470e71d364 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00