[#3124] Delete audio codec when stopping RTP session

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

@@ -267,6 +267,10 @@ namespace sfl {
         delete [] _spkrDataConverted;
         delete _time;
         delete _converter;
+
+        if (_audiocodec) {
+        	delete _audiocodec; _audiocodec = NULL;
+        }
     }
     
     template <typename D>

sflphone-common/src/audio/codecs/codecDescriptor.cpp

@@ -221,8 +221,6 @@ AudioCodec* CodecDescriptor::loadCodec (std::string path) {
 
 void CodecDescriptor::unloadCodec (CodecHandlePointer p) {
 
-	_debug("CodecDescriptor: Unload codec");
-
     using std::cerr;
     destroy_t* destroyCodec = (destroy_t*) dlsym (p.second , "destroy");
 
@@ -250,15 +248,15 @@ AudioCodec* CodecDescriptor::instantiateCodec (AudioCodecType payload) {
             AudioCodec* a = createCodec();
 
             return a;
-
         }
-
         iter++;
     }
 
     return NULL;
 }
 
+
+
 AudioCodec* CodecDescriptor::getFirstCodecAvailable (void) {
 
     CodecsMap::iterator iter = _CodecsMap.begin();

sflphone-common/src/sip/sipvoiplink.cpp

@@ -3276,14 +3276,14 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
 	AudioCodec* audiocodec = Manager::instance().getCodecDescriptorMap().instantiateCodec(pl);
 
 	if (audiocodec == NULL)
-		_error ("SIP: No audiocodec found");
+		_error ("UserAgent: No audiocodec found");
 
 
     try {
         call->setAudioStart (true);
         call->getAudioRtp()->start(audiocodec);
     } catch (exception& rtpException) {
-        _debug ("%s", rtpException.what());
+        _error ("UserAgent: Error: %s", rtpException.what());
     }
 
 }