[#4874] Store call id within rtp thread context

sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp

@@ -53,6 +53,7 @@ AudioRtpRecord::AudioRtpRecord () : _audioCodec (NULL)
     , _spkrAmplFactor (initFadeinFactor)
     , _audioProcess (NULL)
     , _noiseSuppress (NULL)
+    , _callId ("")
 {
 
 }

sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h

@@ -110,6 +110,7 @@ class AudioRtpRecord
         AudioProcessing *_audioProcess;
         NoiseSuppress *_noiseSuppress;
         ost::Mutex audioProcessMutex;
+        std::string _callId;
 
 };
 
@@ -207,10 +208,11 @@ class AudioRtpRecordHandler
 
         void putDtmfEvent (int digit);
 
-    private:
+    protected:
 
         AudioRtpRecord	_audioRtpRecord;
 
+    private:
 
         SIPCall *_ca;
 

sflphone-common/src/audio/audiortp/AudioRtpSession.cpp

@@ -65,6 +65,8 @@ AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
 
     _info ("AudioRtpSession: Setting new RTP session with destination %s:%d", _ca->getLocalIp().c_str(), _ca->getLocalAudioPort());
 
+    _audioRtpRecord._callId = _ca->getCallId();
+
     // static_cast<ost::DualRTPUDPIPv4Channel>(dso)->sendSocket->setTypeOfService(ost::Socket::tosLowDelay);
     // static_cast<ost::DualRTPChannel<ost::DualRTPUDPIPv4Channel> >(dso)->sendSocket->setTypeOfService(ost::Socket::tosLowDelay);
 
@@ -82,7 +84,7 @@ AudioRtpSession::~AudioRtpSession()
         throw;
     }
 
-    Manager::instance().getMainBuffer()->unBindAll (_ca->getCallId());
+    Manager::instance().getMainBuffer()->unBindAll (_audioRtpRecord._callId);
 
     if (_time)
         delete _time;
@@ -362,7 +364,7 @@ void AudioRtpSession::run ()
     // Set recording sampling rate
     _ca->setRecordingSmplRate (getCodecSampleRate());
 
-    _debug ("AudioRtpSession: Entering mainloop for call %s",_ca->getCallId().c_str());
+    _debug ("AudioRtpSession: Entering mainloop for call %s", _audioRtpRecord._callId.c_str());
 
     uint32 timeout = 0;
 
@@ -412,7 +414,7 @@ void AudioRtpSession::run ()
         }
     }
 
-    _debug ("AudioRtpSession: Left main loop for call %s", _ca->getCallId().c_str());
+    _debug ("AudioRtpSession: Left main loop for call %s", _audioRtpRecord._callId.c_str());
 
     Thread::exit();