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:
@ -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
|
||||
|
||||
Reference in New Issue
Block a user