Cleaned up debug messages.

sflphone-common/src/call.cpp

@@ -39,16 +39,12 @@ Call::Call (const CallID& id, Call::CallType type)
     SOUND_FORMAT soundFormat = INT16;
 
     recAudio.setRecordingOption (fileType,soundFormat,44100, Manager::instance().getConfigString (AUDIO, RECORD_PATH),id);
-    // _debug("CALL::Constructor for this clss is called \n");
 }
 
 
 Call::~Call()
 {
-    // _debug("CALL::~Call(): Destructor for this clss is called \n");
-
     if (recAudio.isOpenFile()) {
-        // _debug("CALL::~Call(): A recording file is open, close it \n");
         recAudio.closeFile();
     }
 }

sflphone-common/src/managerimpl.cpp

@@ -350,7 +350,7 @@ ManagerImpl::hangupCall (const CallID& id)
 
     int nbCalls = getCallList().size();
 
-    _debug ("nbCalls %i \n", nbCalls);
+    _debug ("hangupCall: callList is of size %i call(s)\n", nbCalls);
 
     // stop stream
     if (! (nbCalls > 1))
@@ -1422,7 +1422,7 @@ ManagerImpl::setActiveCodecList (const std::vector<  std::string >& list)
     _codecDescriptorMap.saveActiveCodecs (list);
     // setConfig
     std::string s = serialize (list);
-    printf ("%s\n", s.c_str());
+    _debug ("Setting codec with payload number %s to the active list\n", s.c_str());
     setConfig ("Audio", "ActiveCodecs", s);
 }
 
@@ -1467,7 +1467,7 @@ ManagerImpl::serialize (std::vector<std::string> v)
 std::vector <std::string>
 ManagerImpl::getActiveCodecList (void)
 {
-    _debug ("Get Active codecs list\n");
+    _debug ("ManagerImpl::getActiveCodecList\n");
     std::vector< std::string > v;
     CodecOrder active = _codecDescriptorMap.getActiveCodecs();
     unsigned int i=0;
@@ -1477,7 +1477,7 @@ ManagerImpl::getActiveCodecList (void)
         std::stringstream ss;
         ss << active[i];
         v.push_back ( (ss.str()).data());
-        _debug ("%s\n", ss.str().data());
+        _debug ("Codec with payload number %s is active\n", ss.str().data());
         i++;
     }
 
@@ -2709,7 +2709,7 @@ ManagerImpl::loadAccountMap()
         }
 
         if (tmpAccount != NULL) {
-            _debug (" %s \n", iter->c_str());
+            _debug ("Loading account %s \n", iter->c_str());
             _accountMap[iter->c_str() ] = tmpAccount;
             nbAccount++;
         }
@@ -2775,7 +2775,6 @@ ManagerImpl::getAccountIdFromNameAndServer (const std::string& userName, const s
     for (iter = _accountMap.begin(); iter != _accountMap.end(); ++iter) {
         _debug ("for : account = %s\n", iter->first.c_str());
         account = dynamic_cast<SIPAccount *> (iter->second);
-        _debug ("account != NULL = %i\n", (account != NULL));
 
         if (account != NULL) {
             if (account->fullMatch (userName, server)) {

sflphone-common/src/sdp.cpp

@@ -477,7 +477,7 @@ void Sdp::set_local_media_capabilities ()
     // Clean it first
     _local_media_cap.clear();
 
-    _debug ("Fetch local media capabilities .......... %i\n" , get_local_extern_audio_port());
+    _debug ("Fetch local media capabilities. Local extern audio port: %i\n" , get_local_extern_audio_port());
 
     /* Only one audio media used right now */
     audio = new sdpMedia (MIME_TYPE_AUDIO);

sflphone-common/src/sipcall.cpp

@@ -33,7 +33,7 @@ SIPCall::SIPCall (const CallID& id, Call::CallType type, pj_pool_t *pool) : Call
         , _local_sdp (0)
 {
     _local_sdp = new Sdp (pool);
-    _debug ("SIPCALL::Constructor for this clss is called \n");
+    _debug ("SIPCALL::Constructor for this class is called \n");
 }
 
 SIPCall::~SIPCall()
@@ -41,7 +41,7 @@ SIPCall::~SIPCall()
 
     delete _local_sdp;
     _local_sdp = 0;
-    _debug ("SIPCALL::Destructor for this clss is called \n");
+    _debug ("SIPCALL::Destructor for this class is called \n");
 }
 
 

sflphone-common/src/sipvoiplink.cpp

@@ -32,6 +32,28 @@
 
 #define CAN_REINVITE        1
 
+static char * invitationStateMap[] = { 
+"PJSIP_INV_STATE_NULL", 
+"PJSIP_INV_STATE_CALLING", 
+"PJSIP_INV_STATE_INCOMING", 
+"PJSIP_INV_STATE_EARLY", 
+"PJSIP_INV_STATE_CONNECTING", 
+"PJSIP_INV_STATE_CONFIRMED", 
+"PJSIP_INV_STATE_DISCONNECTED" 
+};
+                                       
+static char * transactionStateMap[] = {
+"PJSIP_TSX_STATE_NULL" ,	
+"PJSIP_TSX_STATE_CALLING", 	
+"PJSIP_TSX_STATE_TRYING", 	
+"PJSIP_TSX_STATE_PROCEEDING", 	
+"PJSIP_TSX_STATE_COMPLETED", 	
+"PJSIP_TSX_STATE_CONFIRMED", 	
+"PJSIP_TSX_STATE_TERMINATED", 	
+"PJSIP_TSX_STATE_DESTROYED", 	
+"PJSIP_TSX_STATE_MAX" 
+};
+
 struct result
 {
     pj_status_t             status;
@@ -574,7 +596,7 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
         setCallAudioLocal (call, getLocalIPAddress(), useStun(), getStunServer());
 
         try {
-            _debug ("CREATE NEW RTP SESSION FROM NEWOUTGOINGCALL\n");
+            _debug ("Creating new rtp session in newOutgoingCall\n");
             _audiortp->createNewSession (call);
         } catch (...) {
             _debug ("Failed to create rtp thread from newOutGoingCall\n");
@@ -1154,12 +1176,10 @@ SIPVoIPLink::SIPStartCall (SIPCall* call, const std::string& subject UNUSED)
 
     strTo = getSipTo (call->getPeerNumber(), account->getHostname());
 
-    _debug ("            To: %s\n", strTo.data());
-
     // Generate the from URI
     strFrom = "sip:" + account->getUsername() + "@" + account->getHostname();
 
-    _debug ("              From: %s\n", strFrom.c_str());
+    _debug ("Placing new call: \nTo: %s\nFrom: %s\n", strTo.data(), strFrom.c_str());
 
     // pjsip need the from and to information in pj_str_t format
     pj_strdup2 (_pool, &from, strFrom.data());
@@ -1929,7 +1949,7 @@ void SIPVoIPLink::handle_reinvite (SIPCall *call)
 // This callback is called when the invite session state has changed
 void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
 {
-    _debug ("--------------------- call_on_state_changed --------------------- %i\n", inv->state);
+    _debug ("call_on_state_changed to state %s\n", invitationStateMap[inv->state]);
 
     SIPCall *call;
     AccountID accId;
@@ -2039,7 +2059,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
             link->SIPCallAnswered (call, rdata);
     }
     else if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
-        _debug ("------------------- Call disconnected ---------------------\n");
+        _debug ("Invitation falled in state \"disconnected\".\n");
         _debug ("State: %i, Disconnection cause: %i\n", inv->state, inv->cause);
 
         switch (inv->cause) {
@@ -2082,7 +2102,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
 // This callback is called after SDP offer/answer session has completed.
 void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
 {
-    _debug ("--------------------- call_on_media_update --------------------- \n");
+    _debug ("call_on_media_update\n");
     
     const pjmedia_sdp_session *local_sdp;
     const pjmedia_sdp_session *remote_sdp;
@@ -2140,8 +2160,7 @@ void call_on_forked (pjsip_inv_session *inv, pjsip_event *e)
 
 void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
 {
-
-    _debug ("--------------------- call_on_tsx_changed --------------------- %i\n", tsx->state);
+    _debug ("call_on_tsx_changed to state %s\n", transactionStateMap[tsx->state]);
 
     if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING &&
             pjsip_method_cmp (&tsx->method, &pjsip_refer_method) ==0) {
@@ -2244,7 +2263,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
     userName = std::string (sip_uri->user.ptr, sip_uri->user.slen);
     server = std::string (sip_uri->host.ptr, sip_uri->host.slen);
 
-    std::cout << userName << " ------------------ " << server << std::endl;
+    _debug("mod_on_rx_request: %s@%s\n", userName.c_str(), server.c_str());
 
     // Get the account id of callee from username and server
     account_id = Manager::instance().getAccountIdFromNameAndServer (userName, server);
@@ -2433,7 +2452,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
 
 pj_bool_t mod_on_rx_response (pjsip_rx_data *rdata UNUSED)
 {
-
+    _debug("Mod on rx response");
     return PJ_SUCCESS;
 }