- minor fixes of warnings + counter of dropped packets

src/engine/agent/Agent_Impl.cpp

@@ -535,6 +535,9 @@ void AgentImpl::processGetMediaStats(JsonCpp::Value& request, JsonCpp::Value& an
 #endif
         if (result.exists(SessionInfo_PacketLoss))
             answer["rtp_lost"] = result[SessionInfo_LostRtp].asInt();
+        if (result.exists(SessionInfo_DroppedRtp))
+            answer["rtp_dropped"] = result[SessionInfo_DroppedRtp].asInt();
+
         if (result.exists(SessionInfo_SentRtp))
             answer["rtp_sent"] = result[SessionInfo_SentRtp].asInt();
         if (result.exists(SessionInfo_ReceivedRtp))

src/engine/endpoint/EP_AudioProvider.h

@@ -43,32 +43,32 @@ public:
   void          sendData(PDatagramSocket s, InternetAddress& destination, const void* dataBuffer, unsigned int datasize);
 
   // Updates SDP offer
-  void          updateSdpOffer(resip::SdpContents::Session::Medium& sdp, SdpDirection direction);
+  void          updateSdpOffer(resip::SdpContents::Session::Medium& sdp, SdpDirection direction) override;
 
   // Called by user agent when session is deleted.
-  void          sessionDeleted();
+  void          sessionDeleted() override;
 
   // Called by user agent when session is terminated.
-  void          sessionTerminated();
+  void          sessionTerminated() override;
 
   // Called by user agent when session is started.
-  void          sessionEstablished(int conntype);
+  void          sessionEstablished(int conntype) override;
 
   // Called by user agent to save media socket for this provider
-  void          setSocket(const RtpPair<PDatagramSocket>& p4, const RtpPair<PDatagramSocket>& p6);
+  void          setSocket(const RtpPair<PDatagramSocket>& p4, const RtpPair<PDatagramSocket>& p6) override;
   
   // Called by user agent to get media socket for this provider
-  RtpPair<PDatagramSocket>& socket(int family);
+  RtpPair<PDatagramSocket>& socket(int family) override;
 
   // Called by user agent to process media stream description from remote peer.
   // Returns true if description is processed succesfully. Otherwise method returns false.
   // myAnswer sets if the answer will be sent after.
-  bool          processSdpOffer(const resip::SdpContents::Session::Medium& media, SdpDirection sdpDirection);
+  bool          processSdpOffer(const resip::SdpContents::Session::Medium& media, SdpDirection sdpDirection) override;
 
 
   void          setState(unsigned state) override;
-  unsigned      state();
+  unsigned      state() override;
-  MT::Statistics  getStatistics();
+  MT::Statistics  getStatistics() override;
   MT::PStream   activeStream();
 
   void readFile(const Audio::PWavFileReader& stream, MT::Stream::MediaDirection direction);

src/engine/endpoint/EP_Session.cpp

@@ -478,6 +478,7 @@ void Session::getSessionInfo(Session::InfoOptions options, VariantMap& info)
     info[SessionInfo_ReceivedRtp] = static_cast<int>(stat.mReceivedRtp);
     info[SessionInfo_ReceivedRtcp] = static_cast<int>(stat.mReceivedRtcp);
     info[SessionInfo_LostRtp] = static_cast<int>(stat.mPacketLoss);
+    info[SessionInfo_DroppedRtp] = static_cast<int>(stat.mPacketDropped);
     info[SessionInfo_SentRtp] = static_cast<int>(stat.mSentRtp);
     info[SessionInfo_SentRtcp] = static_cast<int>(stat.mSentRtcp);
     if (stat.mFirstRtpTime)

src/engine/endpoint/EP_Session.h

@@ -66,12 +66,13 @@ enum SessionInfo
   SessionInfo_ReceivedRtp,
   SessionInfo_ReceivedRtcp,
   SessionInfo_LostRtp,
+  SessionInfo_DroppedRtp,
   SessionInfo_Duration,
   SessionInfo_Jitter,
   SessionInfo_Rtt,
   SessionInfo_BitrateSwitchCounter, // It is for AMR codecs only
   SessionInfo_RemotePeer,
-  SessionInfo_SSRC
+  SessionInfo_SSRC,
 };
 
 

src/engine/media/MT_AudioReceiver.cpp

@@ -26,7 +26,7 @@
 using namespace MT;
 
 // ----------------- RtpBuffer::Packet --------------
-RtpBuffer::Packet::Packet(std::shared_ptr<RTPPacket> packet, int timelength, int rate)
+RtpBuffer::Packet::Packet(const std::shared_ptr<RTPPacket>& packet, int timelength, int rate)
     :mRtp(packet), mTimelength(timelength), mRate(rate)
 {
 }
@@ -120,9 +120,8 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
     // New sequence number
     unsigned newSeqno = packet->GetExtendedSequenceNumber();
 
-    for (PacketList::iterator iter = mPacketList.begin(); iter != mPacketList.end(); iter++)
+    for (Packet& p: mPacketList)
     {
-        Packet& p = *iter;
         unsigned seqno = p.rtp()->GetExtendedSequenceNumber();
 
         if (seqno == newSeqno)
@@ -138,6 +137,7 @@ bool RtpBuffer::add(std::shared_ptr<jrtplib::RTPPacket> packet, int timelength,
             minno = seqno;
     }
 
+    // Get amount of available audio (in milliseconds) in jitter buffer
     int available = findTimelength();
 
     if (newSeqno > minno || (available < mHigh))
@@ -188,6 +188,9 @@ RtpBuffer::FetchResult RtpBuffer::fetch(ResultList& rl)
 
         // Erase from packet list
         mPacketList.erase(mPacketList.begin());
+
+        // Increase number in statistics
+        mStat.mPacketDropped++;
     }
 
     if (total < mLow)
@@ -351,7 +354,7 @@ AudioReceiver::~AudioReceiver()
 }
 
 
-bool AudioReceiver::add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec)
+bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec)
 {
     // Increase codec counter
     mStat.mCodecCount[p->GetPayloadType()]++;
@@ -373,20 +376,20 @@ bool AudioReceiver::add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec)
                 codecIter->second = mCodecList.codecAt(codecIndex).create();
     }
 
-    // Return pointer to codec if needed
+    // Return pointer to codec if needed.get()
     if (codec)
         *codec = codecIter->second.get();
 
     if (mStat.mCodecName.empty())
-        mStat.mCodecName = codecIter->second.get()->name();
+        mStat.mCodecName = codecIter->second->name();
 
     // Estimate time length
-    int timelen = 0, payloadLength = p->GetPayloadLength(), ptype = p->GetPayloadType();
+    int time_length = 0, payloadLength = p->GetPayloadLength(), ptype = p->GetPayloadType();
 
     if (!codecIter->second->rtpLength())
-        timelen = codecIter->second->frameTime();
+        time_length = codecIter->second->frameTime();
     else
-        timelen = int(double(payloadLength) / codecIter->second->rtpLength() * codecIter->second->frameTime() + 0.5);
+        time_length = lround(double(payloadLength) / codecIter->second->rtpLength() * codecIter->second->frameTime());
 
     // Process jitter
     mJitterStats.process(p.get(), codecIter->second->samplerate());
@@ -394,19 +397,19 @@ bool AudioReceiver::add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec)
 
     // Check if packet is CNG
     if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
-        timelen = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
+        time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
     else
         // Check if packet is too short from time length side
-        if (timelen < 2)
+        if (time_length < 2)
         {
             // It will cause statistics to report about bad RTP packet
             // I have to replay last packet payload here to avoid report about lost packet
-            mBuffer.add(p, timelen, codecIter->second->samplerate());
+            mBuffer.add(p, time_length, codecIter->second->samplerate());
             return false;
         }
 
     // Queue packet to buffer
-    return mBuffer.add(p, timelen, codecIter->second->samplerate());
+    return mBuffer.add(p, time_length, codecIter->second->samplerate());
 }
 
 void AudioReceiver::processDecoded(Audio::DataWindow& output, int options)

src/engine/media/MT_AudioReceiver.h

@@ -45,7 +45,7 @@ namespace MT
     class Packet
     {
     public:
-      Packet(std::shared_ptr<RTPPacket> packet, int timelen, int rate);
+      Packet(const std::shared_ptr<RTPPacket>& packet, int timelen, int rate);
       std::shared_ptr<RTPPacket> rtp() const;
       int timelength() const;
       int rate() const;
@@ -109,7 +109,7 @@ namespace MT
     
     // Returns false when packet is rejected as illegal. codec parameter will show codec which will be used for decoding.
     // Lifetime of pointer to codec is limited by lifetime of AudioReceiver (it is container).
-    bool add(std::shared_ptr<jrtplib::RTPPacket> p, Codec** codec = nullptr);
+    bool add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** codec = nullptr);
 
     // Returns false when there is no rtp data from jitter
     enum DecodeOptions
@@ -178,6 +178,7 @@ namespace MT
     void initPvqa();
     void updatePvqa(const void* data, int size);
     float calculatePvqaMos(int rate, std::string& report);
+
     std::shared_ptr<Audio::DataWindow> mPvqaBuffer;
 #endif
 

src/engine/media/MT_Statistics.cpp

@@ -40,7 +40,7 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
         int64_t delta = receiveDelta - timestampDelta;
 
         // Update max delta in milliseconds
-        double delta_in_seconds = fabs(double(delta) / rate);
+        float delta_in_seconds = float(fabs(double(delta) / rate));
         if (delta_in_seconds > mMaxDelta)
             mMaxDelta = delta_in_seconds;
 
@@ -56,7 +56,7 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
         mReceiveTimestamp = timestamp;
 
         // And mJitter are in seconds again
-        mJitter.process(mLastJitter.value() / rate);
+        mJitter.process(mLastJitter.value() / float(rate));
     }
 }
 
@@ -67,7 +67,7 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
 Statistics::Statistics()
     :mReceived(0), mSent(0), mReceivedRtp(0), mSentRtp(0),
       mReceivedRtcp(0), mSentRtcp(0), mDuplicatedRtp(0), mOldRtp(0), mIllegalRtp(0),
-      mPacketLoss(0), mJitter(0.0), mAudioTime(0), mSsrc(0)
+      mPacketLoss(0), mJitter(0.0), mAudioTime(0), mSsrc(0), mPacketDropped(0)
 {
     mBitrateSwitchCounter = 0;
     memset(mLoss, 0, sizeof mLoss);
@@ -94,64 +94,11 @@ void Statistics::reset()
     mIllegalRtp = 0;
     mJitter = 0.0;
     mAudioTime = 0;
+    mPacketDropped = 0;
+
     memset(mLoss, 0, sizeof mLoss);
 }
 
-/*
-double calculate_mos_g711(double ppl, double burstr, int version) {
-double r;
-double bpl = 8.47627; //mos = -4.23836 + 0.29873 * r - 0.00416744 * r * r + 0.0000209855 * r * r * r;
-double mos;
-
-if(ppl == 0 or burstr == 0) {
-return 4.5;
-}
-
-if(ppl > 0.5) {
-return 1;
-}
-
-switch(version) {
-case 1:
-case 2:
-default:
-// this mos is calculated for G.711 and PLC
-bpl = 17.2647;
-r = 93.2062077233 - 95.0 * (ppl*100/(ppl*100/burstr + bpl));
-mos = 2.06405 + 0.031738 * r - 0.000356641 * r * r + 2.93143 * pow(10,-6) * r * r * r;
-if(mos < 1)
-return 1;
-if(mos > 4.5)
-return 4.5;
-}
-
-return mos;
-}
-
-
-double calculate_mos(double ppl, double burstr, int codec, unsigned int received) {
-if(codec == PAYLOAD_G729) {
-if(opt_mos_g729) {
-if(received < 100) {
-return 3.92;
-}
-return (double)mos_g729((long double)ppl, (long double)burstr);
-} else {
-if(received < 100) {
-return 4.5;
-}
-return calculate_mos_g711(ppl, burstr, 2);
-}
-} else {
-if(received < 100) {
-return 4.5;
-}
-return calculate_mos_g711(ppl, burstr, 2);
-}
-}
-
-*/
-
 void Statistics::calculateBurstr(double* burstr, double* lossr) const
 {
     int lost = 0;
@@ -230,8 +177,10 @@ Statistics& Statistics::operator += (const Statistics& src)
     mDuplicatedRtp += src.mDuplicatedRtp;
     mOldRtp += src.mOldRtp;
     mPacketLoss += src.mPacketLoss;
+    mPacketDropped += src.mPacketDropped;
     mAudioTime += src.mAudioTime;
 
+
     for (auto codecStat: src.mCodecCount)
     {
         if (mCodecCount.find(codecStat.first) == mCodecCount.end())
@@ -277,6 +226,8 @@ Statistics& Statistics::operator -= (const Statistics& src)
     mDuplicatedRtp -= src.mDuplicatedRtp;
     mOldRtp -= src.mOldRtp;
     mPacketLoss -= src.mPacketLoss;
+    mPacketDropped -= src.mPacketDropped;
+
     mAudioTime -= src.mAudioTime;
     for (auto codecStat: src.mCodecCount)
     {
@@ -291,9 +242,10 @@ Statistics& Statistics::operator -= (const Statistics& src)
 std::string Statistics::toShortString() const
 {
     std::ostringstream oss;
-    oss << "Received: " << mReceivedRtp
+    oss << "Received: "     << mReceivedRtp
-        << ", lost: " << mPacketLoss
+        << ", lost: "       << mPacketLoss
-        << ", sent: " << mSentRtp;
+        << ", dropped: "    << mPacketDropped
+        << ", sent: "       << mSentRtp;
 
     return oss.str();
 }

src/engine/media/MT_Statistics.h

@@ -130,7 +130,9 @@ public:
                 mDuplicatedRtp,   // Number of received duplicated rtp packets
                 mOldRtp,          // Number of late rtp packets
                 mPacketLoss,      // Number of lost packets
+                mPacketDropped,   // Number of dropped packets (due to time unsync when playing)
                 mIllegalRtp;      // Number of rtp packets with bad payload type
+
     int         mLoss[128];       // Every item is number of loss of corresping length
     size_t      mAudioTime;       // Decoded/found time in milliseconds
     uint16_t    mSsrc;            // Last known SSRC ID in a RTP stream