- fixed problem with loss packet counter (it was higher than real value)

src/engine/media/MT_AudioReceiver.cpp

@@ -108,7 +108,7 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
     Lock l(mGuard);
 
     // Update statistics
-    mStat.mSsrc = packet->GetSSRC();
+    mStat.mSsrc = static_cast<uint16_t>(packet->GetSSRC());
 
     // Update jitter
     ICELogMedia(<< "Adding new packet into jitter buffer");
@@ -149,12 +149,14 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
         // Limit by max timelength
         available = findTimelength();
 
-        while (available > mHigh && mPacketList.size())
+        if (available > mHigh)
+            ICELogMedia(<< "Available " << available << "ms with limit " << mHigh << "ms");
+        /*while (available > mHigh && mPacketList.size())
         {
-            //ICELogMedia( << "Dropping RTP packet from jitter");
+            ICELogDebug( << "Dropping RTP packet from jitter buffer");
             available -= mPacketList.front().timelength();
             mPacketList.erase(mPacketList.begin());
-        }
+        }*/
     }
     else
     {
@@ -176,6 +178,18 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
     // See if there is enough information in buffer
     int total = findTimelength();
 
+    while (total > mHigh && mPacketList.size())
+    {
+        ICELogMedia( << "Dropping RTP packets from jitter buffer");
+        total -= mPacketList.front().timelength();
+
+        // Save it as last packet however - to not confuse loss packet counter
+        mFetchedPacket = mPacketList.front();
+
+        // Erase from packet list
+        mPacketList.erase(mPacketList.begin());
+    }
+
     if (total < mLow)
         result = FetchResult::NoPacket;
     else

src/engine/media/MT_AudioStream.cpp

@@ -313,7 +313,9 @@ void AudioStream::dataArrived(PDatagramSocket s, const void* buffer, int length,
       return;
     }
   }
-   
+
+  //ICELogDebug(<< "Packet no: " << RtpHelper::findPacketNo(mReceiveBuffer, receiveLength));
+
   switch (source.family())
   {
   case AF_INET:
@@ -362,7 +364,7 @@ void AudioStream::dataArrived(PDatagramSocket s, const void* buffer, int length,
 
         // Process incoming data packet
         rtpStream->process(packet);
-
+        
         double rtt = mRtpSession.GetCurrentSourceInfo()->INF_GetRoundtripTime().GetDouble();
         if (rtt > 0)
           mStat.mRttDelay.process(rtt);