Record now implemented in Call class

sflphone-gtk/src/actions.c

@@ -259,6 +259,7 @@ sflphone_hang_up()
 			case CALL_STATE_HOLD:
 			case CALL_STATE_BUSY:
 			case CALL_STATE_FAILURE:
+                        case CALL_STATE_RECORDING:
 				dbus_hang_up (selectedCall);
 				selectedCall->state = CALL_STATE_DIALING;
 				(void) time(&selectedCall->_stop);
@@ -330,6 +331,10 @@ sflphone_on_hold ()
 			case CALL_STATE_CURRENT:
 				dbus_hold (selectedCall);
 				break;
+                        case CALL_STATE_RECORD:
+                                dbus_hold (selectedCall);
+                                break;
+                                
 			default:
 				g_warning("Should not happen in sflphone_on_hold!");
 				break;
@@ -714,17 +719,16 @@ sflphone_rec_call()
   {
       case CALL_STATE_CURRENT:
             selectedCall->state = CALL_STATE_RECORD;
-            update_call_tree(current_calls,selectedCall);
-            update_menus();
             break; 
       case CALL_STATE_RECORD:
             selectedCall->state = CALL_STATE_CURRENT;
-            update_call_tree(current_calls,selectedCall);
-            update_menus();
             break;
       default: 
+            g_warning("Should not happen in sflphone_off_hold ()!");
         break;
   }
+  update_call_tree(current_calls,selectedCall);
+  update_menus();
 }
 
 /* Internal to action - set the __CURRENT_ACCOUNT variable */
@@ -761,7 +765,7 @@ sflphone_fill_codec_list()
     c->_bandwidth = atof(details[3]);
     codec_list_add(c);
   }
- 
+
   for(pl=codecs; *codecs; codecs++)
   {
     details = (gchar **)dbus_codec_details(atoi(*codecs));

src/audio/audiortp.cpp

@@ -101,7 +101,8 @@ void
 AudioRtp::setRecording() {
   
   _debug("AudioRtp::setRecording");
-  _RTXThread->recAudio.setRecording();
+  // _RTXThread->recAudio.setRecording();
+  _RTXThread->_ca->setRecording();
   
 }
 
@@ -203,7 +204,11 @@ AudioRtpRTX::initAudioRtpSession (void)
     FILE_TYPE ft = FILE_WAV;
     SOUND_FORMAT sf = INT16;
     recAudio.setSndSamplingRate(44100);
-    recAudio.openFile(_ca->getFileName(),ft,sf,_ca->getCallId());
+    recAudio.openFile(_ca->getFileName(),ft,sf);
+    
+    //_debug("Opening the wave file in call nstance\n");
+    //_ca->recAudio.setSndSamplingRate(44100);
+    //_ca->recAudio.openFile("testRecFromCall",ft,sf,_ca->getCallId());
 
 
     if (!_sym) {
@@ -459,13 +464,17 @@ AudioRtpRTX::run () {
       // Let's wait for the next transmit cycle
 
       recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
-      
+      _ca->recAudio.recData(spkrDataConverted,micData,_nSamplesSpkr,_nSamplesMic);
+
       Thread::sleep(TimerPort::getTimer()); 
       TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
     }
 
     _debug("Close wave file\n");
     recAudio.closeFile();
+   
+    //_debug("Close wave file in call instance\n");
+    //_ca->recAudio.closeFile();
 
     // _debug("stop stream for audiortp loop\n");
     audiolayer->stopStream();
@@ -475,6 +484,9 @@ AudioRtpRTX::run () {
     
     _debug("! Close wave file\n");
     recAudio.closeFile();
+   
+    //_debug("Close wave file in call instance\n");
+    //_ca->recAudio.closeFile();
  
     throw;
   } catch(...) {
@@ -482,6 +494,9 @@ AudioRtpRTX::run () {
     _debugException("* ARTP Action: Stop");
     _debug("* Close wave file\n");
     recAudio.closeFile(); 
+   
+    //_debug("Close wave file in call instance\n");
+    //_ca->recAudio.closeFile();
 
     throw;
   }

src/audio/audiortp.h

@@ -29,7 +29,7 @@
 #include <cc++/numbers.h>
 
 #include "../global.h"
-#include "plug-in/audiorecorder/audiorecord.h"
+// #include "plug-in/audiorecorder/audiorecord.h"
 #include "../samplerateconverter.h"
 
 #define UP_SAMPLING 0
@@ -71,6 +71,9 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
      */
     AudioRecord recAudio;
 
+    /** A SIP call */
+    SIPCall* _ca;    
+
   private:
 
     // copy constructor
@@ -79,9 +82,6 @@ class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
     // assignment operator
     AudioRtpRTX& operator=(const AudioRtpRTX& rh);
 
-    /** A SIP call */
-    SIPCall* _ca;
-
     /** RTP session to send data */
     ost::RTPSession *_sessionSend;
     

src/call.cpp

@@ -55,11 +55,21 @@ Call::Call(const CallID& id, Call::CallType type)
     _filename = out.str();
 
     printf("Call::constructor filename for tis call %s \n",_filename.c_str());
+
+
+
+    FILE_TYPE ft = FILE_WAV;
+    SOUND_FORMAT sf = INT16;
+    _debug("CALL::Opening the wave file in call nstance\n");
+    recAudio.setSndSamplingRate(44100);
+    recAudio.openFile("testRecFromCall",ft,sf);
 }
 
 
 Call::~Call()
 {
+   _debug("CALL::Close wave file in call instance\n");
+   recAudio.closeFile();
 }
 
 void 
@@ -146,3 +156,8 @@ Call::isAudioStarted()
   return _audioStarted;
 }
 
+void
+Call::setRecording()
+{
+  recAudio.setRecording();
+}

src/call.h

@@ -218,7 +218,13 @@ class Call{
     /**
      * A recorder fro this call
      */
-    AudioRecord audioRec; 
+    AudioRecord recAudio;
+  
+    /**
+     * SetRecording
+     */
+    void setRecording();
+     
 
   protected:
     /** Protect every attribute that can be changed by two threads */

src/plug-in/audiorecorder/audiorecord.cpp

@@ -36,7 +36,7 @@ void AudioRecord::setSndSamplingRate(int smplRate){
 }
 
 
-void AudioRecord::openFile(std::string name, FILE_TYPE type, SOUND_FORMAT format, CallID& id){
+void AudioRecord::openFile(std::string name, FILE_TYPE type, SOUND_FORMAT format){
   
    _debug("AudioRecord::openFile()\n");  
   

src/plug-in/audiorecorder/audiorecord.h

@@ -61,7 +61,7 @@ public:
    * @param type     The sound file format (FILE_RAW, FILE_WAVE)
    * @param format   Internal sound format (INT16 / INT32)
    */
-  void openFile(std::string fileName, FILE_TYPE type, SOUND_FORMAT format, CallID& id);
+  void openFile(std::string fileName, FILE_TYPE type, SOUND_FORMAT format);
 
   /**
    * Close the opend recording file. If wave: cout the number of byte