Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2383004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org
2013-11-20 12:46:11 +00:00
parent 27326b6a42
commit 1ae1d0c471
12 changed files with 115 additions and 59 deletions

View File

@ -19,6 +19,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/gtest_prod_util.h"
#ifdef MATLAB
class MatlabPlot;
@ -383,9 +384,13 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
Clock* clock_;
private:
FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
int64_t RtcpReportInterval();
void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
void set_rtt_ms(uint32_t rtt_ms);
uint32_t rtt_ms() const;
int32_t id_;
const bool audio_;
bool collision_detected_;
@ -414,7 +419,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
MatlabPlot* plot1_;
#endif
RtcpRttObserver* rtt_observer_;
RtcpRttStats* rtt_stats_;
// The processed RTT from RtcpRttStats.
scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
uint32_t rtt_ms_;
};
} // namespace webrtc