Règle le problème de transmission audio quand c'est l'appel COURANT (currentCall)

Le tout est synchronisé avec le Manager, pour qu'on traite l'audio
et qu'on prenne le micro seulement quand c'est notre appel qui est en
cours.

src/iaxvoiplink.cpp

@@ -166,13 +166,12 @@ IAXVoIPLink::terminateIAXCall()
 void
 IAXVoIPLink::getEvent() 
 {
-  // mutex here
-  _mutexIAX.enterMutex();
+  IAXCall* call = NULL;
 
+  // lock iax_ stuff..
+  _mutexIAX.enterMutex();
   iax_event* event = NULL;
-  IAXCall* call = NULL;
   while ( (event = iax_get_event(IAX_NONBLOCKING)) != NULL ) {
-
     // If we received an 'ACK', libiax2 tells apps to ignore them.
     if (event->etype == IAX_EVENT_NULL) {
       continue;
@@ -183,31 +182,66 @@ IAXVoIPLink::getEvent()
     call = iaxFindCallBySession(event->session);
 
     if (call) {
-
-      //_debug("  - We've got an associated call, handle call event\n");
+      // We know that call, deal with it
       iaxHandleCallEvent(event, call);
-
-    } else if (event->session && event->session == _regSession) {
-      
-      //_debug("  - We've got an associated REGISTRATION session, handle registration process\n");
-      // in iaxclient, there is many session handling, here, only one
+    }
+    else if (event->session && event->session == _regSession) {
+      // This is a registration session, deal with it
       iaxHandleRegReply(event);
-      
-    } else {
-      
-      //_debug ("  - We've got some other event, deal with them alone.\n");
+    }
+    else {
+      // We've got an event before it's associated with any call
       iaxHandlePrecallEvent(event);
-      
     }
     
     iax_event_free(event);
   }
-
   _mutexIAX.leaveMutex();
 
+
+  // Do the doodle-moodle to send audio from the microphone to the IAX channel.
+  sendAudioFromMic();
+
+
+  // Refresh registration.
+  if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
+    setRegister();
+  }
+
+  // thread wait 5 millisecond
+  _evThread->sleep(5);
+}
+
+
+void
+IAXVoIPLink::sendAudioFromMic(void)
+{
+  IAXCall* currentCall = dynamic_cast<IAXCall*>(getCall(Manager::instance().getCurrentCallId()));
+  AudioCodec* audiocodec = NULL;
+  
+  if (!currentCall) {
+    // Let's mind our own business.
+    return;
+  }
+
+  // Just make sure the currentCall is in state to receive audio right now.
+  if (currentCall->getConnectionState() != Call::Connected) {
+    return;
+  }
+
+  audiocodec = currentCall->getAudioCodec();
+
+  if (!audiocodec) {
+    // Audio codec still not determined.
+    if (audiolayer) {
+      // To keep latency low..
+      audiolayer->flushMic();
+    }
+    return;
+  }
+
   // Send sound here
-  if (_currentCall && audiolayer) {
-    AudioCodec* audiocodec = _currentCall->getAudioCodec();
+  if (audiolayer) {
 
     // we have to get 20ms of data from the mic *20/1000 = /50
     // rate/50 shall be lower than IAX__20S_48KHZ_MAX
@@ -272,24 +306,21 @@ IAXVoIPLink::getEvent()
     // Send it out!
     _mutexIAX.enterMutex();
     // Make sure the session and the call still exists.
-    if (_currentCall->getSession()) {
-      if ( iax_send_voice(_currentCall->getSession(), _currentCall->getFormat(), (unsigned char*)_sendDataEncoded, compSize, nbSample) == -1) {
+    if (currentCall->getSession()) {
+      if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), (unsigned char*)_sendDataEncoded, compSize, nbSample) == -1) {
 	_debug("IAX: Error sending voice data.\n");
       }
     }
     _mutexIAX.leaveMutex();
   }
-
-  // Refresh registration.
-  if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
-    setRegister();
-  }
-
-  // thread wait 5 millisecond
-  _evThread->sleep(5);
 }
 
 
+/*
+void IAXVoIPLink::recvAudioForSpkr(void) 
+{
+}
+*/
 
 
 
@@ -573,7 +604,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
   case IAX_EVENT_HANGUP:
     Manager::instance().peerHungupCall(id); 
     if (Manager::instance().isCurrentCall(id)) {
-      _currentCall = NULL;
       audiolayer->stopStream();
       // stop audio
     }
@@ -584,7 +614,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
     Manager::instance().peerHungupCall(id); 
     if (Manager::instance().isCurrentCall(id)) {
       // stop audio
-      _currentCall = NULL;
       audiolayer->stopStream();
     }
     removeCall(id);
@@ -609,7 +638,6 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
       }
       
       Manager::instance().peerAnsweredCall(id);
-      _currentCall = call;
       audiolayer->startStream();
       // start audio here?
     } else {

src/iaxvoiplink.h

@@ -129,6 +129,11 @@ private:
    */
   void iaxHandlePrecallEvent(iax_event* event);
 
+  /**
+   * Work out the audio data from Microphone to IAX2 channel
+   */
+  void sendAudioFromMic(void);
+
   /**
    * Send an outgoing call invite to iax
    * @param call An IAXCall pointer
@@ -199,7 +204,7 @@ private:
   int           _src_err;
 
   /** Current IAX call pointer, used for sending, change when starting audio, switching */
-  IAXCall* _currentCall; 
+  //IAXCall* _currentCall; 
 };
 
 #endif