Hold/unhold : quelch/unquelch and pointer error fix

ChangeLog

+2006-08-02  Yan Morin
+
+	* Add IAX quelch/unquelch
+
+	* Add SAMPLERATE for incoming voice data
+
 2006-07-27  Yan Morin
 
 	* Add IAX registration and outgoing call

src/Makefile.am

@@ -25,6 +25,14 @@ IAXSOURCES =
 IAXHEADERS =
 endif
 
+if USE_SAMPLERATE
+SAMPLERATE_FLAG=-DUSE_SAMPLERATE
+SAMPLERATE_LIB =-lsamplerate
+else
+SAMPLERATE_FLAG=
+SAMPLERATE_LIB=
+endif
+
 SUBDIRS = audio config gui $(ZEROCONFDIR)
 
 sflphoned_SOURCES = eventthread.cpp 	main.cpp 	voIPLink.cpp \
@@ -37,7 +45,7 @@ sflphoned_CXXFLAGS = -DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone
 
 #sflphoned_LDFLAGS = -static
 sflphoned_LDADD =  ./libsflphone.la $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) $(EXOSIP_LIBS)
-AM_CPPFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $(libccgnu2_CFLAGS) $(IAX_CFLAGS) $(USER_INCLUDES)
+AM_CPPFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $(libccgnu2_CFLAGS) $(IAX_CFLAGS) $(USER_INCLUDES) $(SAMPLERATE_FLAG)
 
 #libsflphone_la_LDFLAGS= -version-info 0:1:0
 libsflphone_la_LIBADD = \
@@ -46,7 +54,7 @@ libsflphone_la_LIBADD = \
 	./audio/libaudio.la \
 	./gui/libguiframework.la \
 	./config/libconfig.la \
-	$(IAX_LIBS)
+	$(IAX_LIBS) $(SAMPLERATE_LIB)
 
 libsflphone_la_SOURCES =
 

src/audio/audiortp.cpp

@@ -350,32 +350,8 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers_stereo, int16* data
         src_data.output_frames = _nbFrames;
         src_data.src_ratio = factord;
         src_short_to_float_array(data_for_speakers_recv, _floatBufferIn, nbInt16);
-#if 0
-        for(int i=0; i<20; i++) {
-          printf("%3d ", data_for_speakers_recv[i]);
-        }
-        printf("\n");
-        for(int i=0; i<20; i++) {
-          printf("%7.6f ", _floatBufferIn[i]);
-        }
-        printf("\n");
-#endif
         src_simple (&src_data, SRC_SINC_MEDIUM_QUALITY, fromChannel);
         src_float_to_short_array(_floatBufferOut, data_for_speakers_recv, src_data.output_frames_gen*fromChannel);
-
-#if 0
-        for(int i=0; i<100; i++) {
-          printf("%7.6f ", _floatBufferOut[i]);
-        }
-        printf("\n");
-        for(int i=0; i<100; i++) {
-          printf("%3d ", data_for_speakers_recv[i]);
-        }
-        printf("\n");
-
-        printf("receive get %d and convert to %d\n", nbInt16, src_data.output_frames_gen);
-#endif
-
         nbInt16 = src_data.output_frames_gen*fromChannel;
       }
 #endif

src/iaxvoiplink.cpp

@@ -25,6 +25,10 @@
 #include "audio/audiocodec.h"
 #include "audio/audiolayer.h"
 
+#ifdef USE_SAMPLERATE
+ #include <samplerate.h>
+#endif
+
 #define IAX_SUCCESS  0
 #define IAX_FAILURE -1
 
@@ -67,9 +71,14 @@ IAXVoIPLink::IAXVoIPLink(const AccountID& accountID)
 
   audiocodec = 0;
   audiolayer = 0;
- _nbFrames = 160;
+  _nbFrames = 160;
   data_for_speakers_recv = new int16[_nbFrames*2];
   data_for_speakers_output = new int16[_nbFrames*2];
+
+#ifdef USE_SAMPLERATE
+  _floatBufferIn  = new float[_nbFrames*2];
+  _floatBufferOut = new float[_nbFrames*2];
+#endif
 }
 
 
@@ -83,6 +92,11 @@ IAXVoIPLink::~IAXVoIPLink()
   audiolayer = 0;
   delete [] data_for_speakers_recv; data_for_speakers_recv = 0;
   delete [] data_for_speakers_output; data_for_speakers_output = 0;
+
+#ifdef USE_SAMPLERATE
+  delete [] _floatBufferIn;
+  delete [] _floatBufferOut;
+#endif
 }
 
 bool
@@ -130,9 +144,30 @@ IAXVoIPLink::terminate()
 {
 //  iaxc_shutdown();  
 //  hangup all call
+    terminateIAXCall();
 //  iax_hangup(calls[callNo].session,"Dumped Call");
 }
 
+void
+IAXVoIPLink::terminateIAXCall()
+{
+  ost::MutexLock m(_callMapMutex);
+  CallMap::iterator iter = _callMap.begin();
+  IAXCall *call;
+  while( iter != _callMap.end() ) {
+    call = dynamic_cast<IAXCall*>(iter->second);
+    if (call) {
+      _mutexIAX.enterMutex();
+      iax_hangup(call->getSession(),"Dumped Call");
+      _mutexIAX.leaveMutex();
+      call->setSession(0);
+      delete call; call = 0;
+    }
+    iter++;
+  }
+  _callMap.clear();
+}
+
 void
 IAXVoIPLink::getEvent() 
 {
@@ -273,6 +308,33 @@ IAXVoIPLink::hangup(const CallID& id)
 	removeCall(id);
 	return true;	
 }
+
+bool 
+IAXVoIPLink::onhold(const CallID& id) 
+{
+  IAXCall* call = getIAXCall(id);
+  if (call==0) { _debug("Call doesn't exists\n"); return false; }
+  //if (call->getState() == Call::Hold) { _debug("Call is already on hold\n"); return false; }
+  _mutexIAX.enterMutex();
+  iax_quelch(call->getSession());
+  _mutexIAX.leaveMutex();
+  call->setState(Call::Hold);
+  return true;
+}
+
+bool 
+IAXVoIPLink::offhold(const CallID& id)
+{
+  IAXCall* call = getIAXCall(id);
+  if (call==0) { _debug("Call doesn't exists\n"); return false; }
+  //if (call->getState() == Call::Active) { _debug("Call is already active\n"); return false; }
+  _mutexIAX.enterMutex();
+  iax_unquelch(call->getSession());
+  _mutexIAX.leaveMutex();
+  call->setState(Call::Active);
+  return true;
+}
+
 bool
 IAXVoIPLink::iaxOutgoingInvite(IAXCall* call) 
 {
@@ -337,11 +399,13 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
   // note activity?
   //
   CallID id = call->getCallId();
+  int16* output = 0; // for audio output
+
   switch(event->etype) {
     case IAX_EVENT_HANGUP:
       Manager::instance().peerHungupCall(id); 
       if (Manager::instance().isCurrentCall(id)) {
-	audiolayer->stopStream();
+        audiolayer->stopStream();
         // stop audio
       }
       removeCall(id);
@@ -351,7 +415,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
       Manager::instance().peerHungupCall(id); 
       if (Manager::instance().isCurrentCall(id)) {
         // stop audio
-	audiolayer->stopStream();
+        audiolayer->stopStream();
       }
       removeCall(id);
      break;
@@ -387,28 +451,48 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
     break;
     
     case IAX_EVENT_VOICE:
+     output = data_for_speakers_output; // don't use data directly, use a pointer instead!!!
+     // since convert return doesn't copy anything if it doesn't have to.
+
      if (audiocodec != 0 && audiolayer!=0) {
-	_debug("Receive: len=%d, format=%d, data_for_speakers_recv=%p\n", event->datalen, call->getFormat(), data_for_speakers_recv);
-	unsigned char* data = (unsigned char*)event->data;
-	unsigned int size   = event->datalen;
+        //_debug("Receive: len=%d, format=%d, data_for_speakers_recv=%p\n", event->datalen, call->getFormat(), data_for_speakers_recv);
+        unsigned char* data = (unsigned char*)event->data;
+        unsigned int size   = event->datalen;
         int expandedSize = audiocodec->codecDecode(data_for_speakers_recv, data, size);
-	int nbInt16      = expandedSize/sizeof(int16);
-	int toChannel = 1;
-	int fromChannel = 1;
-	int toPut;
-        toPut = audiolayer->convert(data_for_speakers_recv, fromChannel, nbInt16, &data_for_speakers_output, toChannel);
-        audiolayer->putMain(data_for_speakers_output, toPut * sizeof(int16));
-
-	_debug("sending %d int16 to speakers..\n", toPut);
+        int nbInt16      = expandedSize/sizeof(int16);
+        int toChannel = 1;
+        int fromChannel = 1;
+        int toPut;
+
+#ifdef USE_SAMPLERATE
+      if ( audiolayer->getSampleRate() != audiocodec->getClockRate() ) {
+        // convert here
+        double         factord = (double)audiolayer->getSampleRate()/ audiocodec->getClockRate();
+
+        SRC_DATA src_data;
+        src_data.data_in = _floatBufferIn;
+        src_data.data_out = _floatBufferOut;
+        src_data.input_frames = nbInt16/fromChannel;
+        src_data.output_frames = _nbFrames;
+        src_data.src_ratio = factord;
+        src_short_to_float_array(data_for_speakers_recv, _floatBufferIn, nbInt16);
+        src_simple (&src_data, SRC_SINC_MEDIUM_QUALITY, fromChannel);
+        src_float_to_short_array(_floatBufferOut, data_for_speakers_recv, src_data.output_frames_gen*fromChannel);
+        nbInt16 = src_data.output_frames_gen*fromChannel;
+      }
+#endif
+        toPut = audiolayer->convert(data_for_speakers_recv, fromChannel, nbInt16, &output, toChannel);
+        audiolayer->putMain(output, toPut * sizeof(int16));
+        //_debug("sending %d int16 to speakers..\n", toPut);
       }
     break;
-    
+
     case IAX_EVENT_TEXT:
     break;
-    
+
     case IAX_EVENT_RINGA:
     break;
-    
+
     case IAX_EVENT_PONG:
     break;
     

src/iaxvoiplink.h

@@ -55,8 +55,8 @@ public:
 
   bool hangup(const CallID& id);
   bool cancel(const CallID& id) { return false; }
-  bool onhold(const CallID& id) { return false; }
-  bool offhold(const CallID& id) { return false; }
+  bool onhold(const CallID& id);
+  bool offhold(const CallID& id);
   bool transfer(const CallID& id, const std::string& to) { return false; }
   bool refuse (const CallID& id) { return false; }
   bool carryingDTMFdigits(const CallID& id, char code) { return false; }
@@ -75,7 +75,12 @@ private:
    */
   IAXCall* getIAXCall(const CallID& id);
 
-  /*
+  /**
+   * Delete every call 
+   */
+  void terminateIAXCall();
+
+  /**
    * Find a iaxcall by iax session number
    * @param session an iax_session valid pointer
    * @return iaxcall or 0 if not found
@@ -119,6 +124,10 @@ private:
   int16* data_for_speakers_recv;
   int16* data_for_speakers_output;
   int _nbFrames;
+#ifdef USE_SAMPLERATE
+  float *_floatBufferIn;
+  float *_floatBufferOut;
+#endif
 };
 
 #endif

src/managerimpl.cpp

@@ -1718,8 +1718,10 @@ ManagerImpl::loadAccountMap()
 void
 ManagerImpl::unloadAccountMap()
 {
+  _debug("Unloading account map...\n");
   AccountMap::iterator iter = _accountMap.begin();
   while ( iter != _accountMap.end() ) {
+    _debug("-> Deleting account %s\n", iter->first.c_str());
     delete iter->second; iter->second = 0;
     iter++;
   }

src/sipvoiplink.cpp

@@ -735,6 +735,7 @@ SIPVoIPLink::offhold(const CallID& id)
 
   // Enable audio
   _debug("SIP: Starting AudioRTP when offhold\n");
+  call->setState(Call::Active);
   // it's sure that this is the current call id...
   if (_audiortp.createNewSession(call) < 0) {
     _debug("SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);

tools/download.sh

@@ -57,7 +57,7 @@ prepare()
     fi
 }
 
-#prepare $SFL_FILE_CCPP2 ''
+prepare $SFL_FILE_CCPP2 ''
 prepare $SFL_FILE_CCRTP ''
 #prepare $SFL_FILE_LIBOSIP2 ''
 #prepare $SFL_FILE_LIBEXOSIP2 ''