Add logs when no RTCP RR has been received for three regular RTCP intervals.

BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 17 insertions, 0 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 5dc0310209..455c0f8079 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -83,6 +83,16 @@ public:
     WebRtc_Word32 StatisticsReceived(
         std::vector<RTCPReportBlock>* receiveBlocks) const;
 
+    // Returns true if we haven't received an RTCP RR for several RTCP
+    // intervals, but only triggers true once.
+    bool RtcpRrTimeout(int64_t rtcp_interval_ms);
+
+    // Returns true if we haven't received an RTCP RR telling the receive side
+    // has not received RTP packets for too long, i.e. extended highest sequence
+    // number hasn't increased for several RTCP intervals. The function only
+    // returns true once until a new RR is received.
+    bool RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms);
+
     // Get TMMBR
     WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
                                 const WebRtc_UWord32 accNumCandidates,
@@ -218,6 +228,13 @@ protected:
 
   WebRtc_UWord32            _packetTimeOutMS;
 
+  // The last time we received an RTCP RR.
+  int64_t _lastReceivedRrMs;
+
+  // The time we last received an RTCP RR telling we have ssuccessfully
+  // delivered RTP packet to the remote side.
+  int64_t _lastIncreasedSequenceNumberMs;
+
   // Externally set RTT. This value can only be used if there are no valid
   // RTT estimates.
   WebRtc_UWord16 _rtt;