- ongoing work to ease jitter & RTT calculation

src/engine/helper/HL_StreamState.h

@@ -1,4 +1,4 @@
-/* Copyright(C) 2007-2016 VoIP objects (voipobjects.com)
+/* Copyright(C) 2007-2023 VoIP objects (voipobjects.com)
  * This Source Code Form is subject to the terms of the Mozilla Public
  * License, v. 2.0. If a copy of the MPL was not distributed with this
  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */

src/engine/media/MT_AudioStream.cpp

@@ -22,9 +22,9 @@
 
 using namespace MT;
 AudioStream::AudioStream(const CodecList::Settings& settings)
-:mPacketTime(0), mEncodedTime(0), mCodecSettings(settings),
+    :mPacketTime(0), mEncodedTime(0), mCodecSettings(settings),
-mRemoteTelephoneCodec(0), mRtpSession(), mTransmittingPayloadType(-1),
+      mRemoteTelephoneCodec(0), mRtpSession(), mTransmittingPayloadType(-1),
-mRtpSender(mStat)
+      mRtpSender(mStat)
 {
     mOutputBuffer.setCapacity(16384);
     mCapturedAudio.setCapacity(16384);
@@ -296,9 +296,9 @@ void AudioStream::dataArrived(PDatagramSocket s, const void* buffer, int length,
     assert(info);
 
     // Drop RTP packets if stream is not receiving now; let RTCP go
-  if (!(state() & (int)StreamState::Receiving) && RtpHelper::isRtp(buffer, length))
+    if (!(state() & (int)StreamState::Receiving) && RtpHelper::isRtpOrRtcp(buffer, length))
     {
-    ICELogMedia(<< "Stream is not allowed to receive RTP stream. Ignore the packet");
+        ICELogMedia(<< "Stream is not allowed to receive RTP stream. Ignore the RT(C)P packet");
         return;
     }
 
@@ -320,8 +320,6 @@ void AudioStream::dataArrived(PDatagramSocket s, const void* buffer, int length,
         }
     }
 
-  //ICELogDebug(<< "Packet no: " << RtpHelper::findPacketNo(mReceiveBuffer, receiveLength));
-
     switch (source.family())
     {
     case AF_INET:
@@ -360,6 +358,7 @@ void AudioStream::dataArrived(PDatagramSocket s, const void* buffer, int length,
         if (packet)
         {
             ICELogMedia(<< "jrtplib returned packet");
+
             // Find right handler for rtp stream
             SingleAudioStream* rtpStream = nullptr;
             AudioStreamMap::iterator streamIter = mStreamMap.find(packet->GetSSRC());

src/engine/media/MT_AudioStream.h

@@ -26,9 +26,9 @@
 namespace MT
 {
 
-  class AudioStream: public Stream
+class AudioStream: public Stream
-  {
+{
-  public:
+public:
     AudioStream(const CodecList::Settings& codecSettings);
     ~AudioStream();
 
@@ -59,7 +59,7 @@ namespace MT
 
     void setFinalStatisticsOutput(Statistics* stats);
 
-  protected:
+protected:
     Audio::DataWindow mCapturedAudio;     // Data from microphone
     Audio::DataWindow mStereoCapturedAudio;
     char mIncomingPcmBuffer[AUDIO_MIC_BUFFER_SIZE]; // Temporary buffer to allow reading from file
@@ -106,7 +106,7 @@ namespace MT
     Statistics* mFinalStatistics = nullptr;
 
     bool decryptSrtp(void* data, int* len);
-  };
+};
 };
 
 #endif

src/engine/media/MT_Box.cpp

@@ -40,7 +40,7 @@ PStream Terminal::createStream(int type, VariantMap& /*config*/)
   switch (type)
   {
   case Stream::Audio:
-    result = PStream(new AudioStream(MT::CodecList::Settings::DefaultSettings));
+    result = std::make_shared<AudioStream>(MT::CodecList::Settings::DefaultSettings);
     mAudioList.add(result);
     break;
 

src/libs/jrtplib/src/rtcpscheduler.cpp

@@ -100,6 +100,7 @@ RTCPScheduler::~RTCPScheduler()
 
 void RTCPScheduler::Reset()
 {
+    pmembers = 0;
 	headeroverhead = 0; // user has to set this to an appropriate value
 	hassentrtcp = false;
 	firstcall = true;

src/libs/jrtplib/src/rtpexternaltransmitter.cpp

@@ -900,9 +900,26 @@ void RTPExternalTransmitter::InjectRTPorRTCP(const void *data, size_t len, const
 	AbortWaitInternal();
 
 	MAINMUTEX_UNLOCK
-
 }
 
+void RTPExternalTransmitter::InjectRaw(RTPRawPacket* packet)
+{
+    if (!init)
+        return;
+
+    MAINMUTEX_LOCK
+    if (!created)
+    {
+        MAINMUTEX_UNLOCK
+        return;
+    }
+    rawpacketlist.push_back(packet);
+    AbortWaitInternal();
+
+    MAINMUTEX_UNLOCK
+}
+
+
 #ifdef RTPDEBUG
 void RTPExternalTransmitter::Dump()
 {

src/libs/jrtplib/src/rtpexternaltransmitter.h

@@ -96,6 +96,8 @@ public:
 
 	/** Use this function to inject an RTP or RTCP packet and the transmitter will try to figure out which type of packet it is. */
 	void InjectRTPorRTCP(const void *data, size_t len, const RTPAddress &a);
+
+    void InjectRaw(RTPRawPacket* packet);
 private:
 	RTPExternalTransmitter *transmitter;
 };
@@ -186,6 +188,8 @@ public:
 	void InjectRTP(const void *data, size_t len, const RTPAddress &a);
 	void InjectRTCP(const void *data, size_t len, const RTPAddress &a);
 	void InjectRTPorRTCP(const void *data, size_t len, const RTPAddress &a);
+    void InjectRaw(RTPRawPacket* packet);
+
 private:
 	void FlushPackets();
 	
@@ -233,6 +237,11 @@ inline void RTPExternalPacketInjecter::InjectRTPorRTCP(const void *data, size_t
 	transmitter->InjectRTPorRTCP(data, len, a); 
 }
 
+inline void RTPExternalPacketInjecter::InjectRaw(RTPRawPacket* packet)
+{
+    transmitter->InjectRaw(packet);
+}
+
 } // end namespace
 
 #endif // RTPTCPSOCKETTRANSMITTER_H

src/libs/jrtplib/src/rtpsession.cpp

@@ -1444,8 +1444,8 @@ int RTPSession::ProcessPolledData()
 	RTPTime colltimeout = RTPTime(Td*collisionmultiplier);
 	RTPTime notetimeout = RTPTime(Td*notemultiplier);
 	
-	sources.MultipleTimeouts(t,sendertimeout,byetimeout,generaltimeout,notetimeout);
+    // sources.MultipleTimeouts(t,sendertimeout,byetimeout,generaltimeout,notetimeout);
-	collisionlist.Timeout(t,colltimeout);
+    // collisionlist.Timeout(t,colltimeout);
 	
 	// We'll check if it's time for RTCP stuff
 

src/libs/jrtplib/src/rtpsession.h

@@ -55,6 +55,8 @@
 	#include <jthread/jmutex.h>	
 #endif // RTP_SUPPORT_THREAD
 
+#include <iostream>
+
 namespace jrtplib
 {
 
@@ -474,7 +476,11 @@ protected:
 	                              const uint8_t *cname,size_t cnamelength)				{ }
 
 	/** Is called when a new entry \c srcdat is added to the source table. */
-	virtual void OnNewSource(RTPSourceData *srcdat)			 				{ }
+    virtual void OnNewSource(RTPSourceData *srcdat)
+    {
+        // Sync timestamp unit
+        srcdat->SetTimestampUnit(timestampunit);
+    }
 
 	/** Is called when the entry \c srcdat is about to be deleted from the source table. */
 	virtual void OnRemoveSource(RTPSourceData *srcdat)						{ }

src/libs/jrtplib/src/rtpsourcedata.cpp

@@ -35,12 +35,12 @@
 #include "rtpaddress.h"
 #include "rtpmemorymanager.h"
 #if ! (defined(WIN32) || defined(_WIN32_WCE))
-	#include <netinet/in.h>
+#include <netinet/in.h>
 #endif // WIN32
 
 #ifdef RTPDEBUG
-	#include <iostream>
+#include <iostream>
-	#include <string>
+#include <string>
 #endif // RTPDEBUG
 
 #include "rtpdebug.h"
@@ -53,7 +53,7 @@
     numnewpackets++;								\
     \
     if (pack->GetExtendedSequenceNumber() == 0)					\
-		{										\
+{										\
     baseseqnr = 0x0000FFFF;							\
     numcycles = 0x00010000;							\
     }										\
@@ -76,7 +76,7 @@ namespace jrtplib
 {
 
 void RTPSourceStats::ProcessPacket(RTPPacket *pack,const RTPTime &receivetime,double tsunit,
-                                   bool ownpacket,bool *accept,bool applyprobation,bool *onprobation)
+                                   bool ownpacket, bool *accept, bool applyprobation, bool *onprobation)
 {
     // Note that the sequence number in the RTP packet is still just the
     // 16 bit number contained in the RTP header
@@ -139,7 +139,7 @@ void RTPSourceStats::ProcessPacket(RTPPacket *pack,const RTPTime &receivetime,do
 
         ACCEPTPACKETCODE
 
-#endif // RTP_SUPPORT_PROBATION
+        #endif // RTP_SUPPORT_PROBATION
     }
     else // already got packets
     {
@@ -197,15 +197,15 @@ void RTPSourceStats::ProcessPacket(RTPPacket *pack,const RTPTime &receivetime,do
             djitter += diff;
             jitter = (uint32_t)djitter;
 #else
-RTPTime curtime = receivetime;
+            RTPTime curtime = receivetime;
-double diffts1,diffts2,diff;
+            double diffts1,diffts2,diff;
-uint32_t curts = pack->GetTimestamp();
+            uint32_t curts = pack->GetTimestamp();
 
-curtime -= prevpacktime;
+            curtime -= prevpacktime;
-diffts1 = curtime.GetDouble()/tsunit;	
+            diffts1 = curtime.GetDouble() / tsunit;
 
-if (curts > prevtimestamp)
+            if (curts > prevtimestamp)
-{
+            {
                 uint32_t unsigneddiff = curts - prevtimestamp;
 
                 if (unsigneddiff < 0x10000000) // okay, curts realy is larger than prevtimestamp
@@ -217,9 +217,9 @@ if (curts > prevtimestamp)
                     unsigneddiff = -unsigneddiff; // to get the actual difference (in absolute value)
                     diffts2 = -((double)unsigneddiff);
                 }
-}
+            }
-else if (curts < prevtimestamp)
+            else if (curts < prevtimestamp)
-{
+            {
                 uint32_t unsigneddiff = prevtimestamp - curts;
 
                 if (unsigneddiff < 0x10000000) // okay, curts really is smaller than prevtimestamp
@@ -231,17 +231,17 @@ else if (curts < prevtimestamp)
                     unsigneddiff = -unsigneddiff; // to get the actual difference (in absolute value)
                     diffts2 = (double)unsigneddiff;
                 }
-}
+            }
-else
+            else
                 diffts2 = 0;
 
-diff = diffts1 - diffts2;
+            diff = diffts1 - diffts2;
-if (diff < 0)
+            if (diff < 0)
                 diff = -diff;
-diff -= djitter;
+            diff -= djitter;
-diff /= 16.0;
+            diff /= 16.0;
-djitter += diff;
+            djitter += diff;
-jitter = (uint32_t)djitter;
+            jitter = (uint32_t)djitter;
 #endif
         }
         else

src/libs/jrtplib/src/rtpsources.cpp

@@ -809,6 +809,8 @@ int RTPSources::ObtainSourceDataInstance(uint32_t ssrc,RTPInternalSourceData **s
 #endif // RTP_SUPPORT_PROBATION
 		if (srcdat2 == 0)
 			return ERR_RTP_OUTOFMEM;
+
+        // Add new source item
         if ((status = sourcelist.AddElement(ssrc,srcdat2)) < 0)
 		{
 			RTPDelete(srcdat2,GetMemoryManager());