Merge branch 'recording'

Conflicts:

	src/audio/audiortp.cpp
	src/sipvoiplink.cpp

src/audio/codecs/speexcodec.cpp

@@ -22,6 +22,7 @@
 #include <cstdio>
 #include <speex/speex.h>
 
+
 class Speex : public AudioCodec{
 public:
 	Speex(int payload=0)

src/call.cpp

@@ -173,6 +173,12 @@ Call::setRecording()
   recAudio.setRecording();
 }
 
+void
+Call::stopRecording()
+{
+  recAudio.stopRecording();
+}
+
 bool
 Call::isRecording()
 {

src/call.h

@@ -224,6 +224,11 @@ class Call{
      * SetRecording
      */
     void setRecording();
+
+    /**
+     * stopRecording, make sure the recording is stopped (whe transfering call)
+     */
+    void stopRecording();
     
     /**
      * Return Recording state

src/iaxvoiplink.cpp

@@ -163,6 +163,17 @@ IAXVoIPLink::terminateIAXCall()
     _callMap.clear();
 }
 
+void IAXVoIPLink::terminateOneCall(const CallID& id)
+{
+    IAXCall* call = getIAXCall(id);
+    if(call){
+        _debug("IAXVoIPLink::terminateOneCall()::the call is deleted, should close recording file \n");
+        delete call; call = 0;
+    }
+}
+
+
+
     void
 IAXVoIPLink::getEvent() 
 {
@@ -207,6 +218,8 @@ IAXVoIPLink::getEvent()
     if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
         sendRegister("");
     }
+    if(call)
+      call->recAudio.recData(spkrDataConverted,micData,nbSample_,nbSample_);
 
     // thread wait 3 millisecond
     _evThread->sleep(3);
@@ -216,7 +229,7 @@ IAXVoIPLink::getEvent()
     void
 IAXVoIPLink::sendAudioFromMic(void)
 {
-    int maxBytesToGet, availBytesFromMic, bytesAvail, nbSample, compSize;
+    int maxBytesToGet, availBytesFromMic, bytesAvail, compSize;
     AudioCodec *ac;
 
     // We have to update the audio layer type in case we switched
@@ -273,19 +286,19 @@ IAXVoIPLink::sendAudioFromMic(void)
         //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
 
         // Get bytes from micRingBuffer to data_from_mic
-        nbSample = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
+        nbSample_ = audiolayer->getMic( micData, bytesAvail ) / sizeof(SFLDataFormat);
 
         // resample
-        nbSample = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() ,  (int)audiolayer->getSampleRate() , nbSample );
+        nbSample_ = converter->downsampleData( micData , micDataConverted , (int)ac ->getClockRate() ,  (int)audiolayer->getSampleRate() , nbSample_ );
 
         // for the mono: range = 0 to IAX_FRAME2SEND * sizeof(int16)
-        compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample*sizeof(int16));
+        compSize = ac->codecEncode( micDataEncoded, micDataConverted , nbSample_*sizeof(int16));
 
         // 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(), micDataEncoded, compSize, nbSample) == -1) {
+            if (iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample_) == -1) {
                 _debug("IAX: Error sending voice data.\n");
             }
         }
@@ -421,11 +434,12 @@ IAXVoIPLink::answer(const CallID& id)
     bool 
 IAXVoIPLink::hangup(const CallID& id)
 {
+    _debug("IAXVoIPLink::hangup() : function called once hangup \n");
     IAXCall* call = getIAXCall(id);
     std::string reason = "Dumped Call";
     CHK_VALID_CALL;
-
     _mutexIAX.enterMutex();
+    
     iax_hangup(call->getSession(), (char*) reason.c_str());
     _mutexIAX.leaveMutex();
     call->setSession(NULL);
@@ -433,10 +447,34 @@ IAXVoIPLink::hangup(const CallID& id)
         // stop audio
         audiolayer->stopStream();
     }
+    terminateOneCall(id);
+    removeCall(id);
+    return true;	
+}
+
+
+bool 
+IAXVoIPLink::peerHungup(const CallID& id)
+{
+    _debug("IAXVoIPLink::peerHangup() : function called once hangup \n");
+    IAXCall* call = getIAXCall(id);
+    std::string reason = "Dumped Call";
+    CHK_VALID_CALL;
+    _mutexIAX.enterMutex();
+    
+    _mutexIAX.leaveMutex();
+    call->setSession(NULL);
+    if (Manager::instance().isCurrentCall(id)) {
+        // stop audio
+        audiolayer->stopStream();
+    }
+    terminateOneCall(id);
     removeCall(id);
     return true;	
 }
 
+
+
     bool 
 IAXVoIPLink::onhold(const CallID& id) 
 {
@@ -502,6 +540,8 @@ IAXVoIPLink::refuse(const CallID& id)
     iax_reject(call->getSession(), (char*) reason.c_str());
     _mutexIAX.leaveMutex();
 
+
+    // terminateOneCall(id);
     removeCall(id);
 
     return true;
@@ -511,13 +551,21 @@ IAXVoIPLink::refuse(const CallID& id)
 void 
 IAXVoIPLink::setRecording(const CallID& id)
 {
-  _debug("IAXVoIPLink::setRecording!");
+  _debug("IAXVoIPLink::setRecording()!");
+
+  IAXCall* call = getIAXCall(id);
+
+  call->setRecording();
 }
 
 bool 
 IAXVoIPLink::isRecording(const CallID& id)
 {
-  _debug("IAXVoIPLink::setRecording!");
+  _debug("IAXVoIPLink::setRecording()!");
+
+  IAXCall* call = getIAXCall(id);
+
+  return call->isRecording();
 }
 
 
@@ -604,7 +652,17 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
                 // stop audio
                 audiolayer->stopStream();
             }
-            Manager::instance().peerHungupCall(id); 
+            Manager::instance().peerHungupCall(id);
+            /*
+            _debug("IAXVoIPLink::iaxHandleCallEvent, peer hangup have been called");        
+            std::string reason = "Dumped Call";
+            _mutexIAX.enterMutex();
+            iax_hangup(call->getSession(), (char*)reason.c_str());
+            _mutexIAX.leaveMutex();
+            call->setSession(NULL);
+            audiolayer->stopStream();
+            terminateOneCall(id);
+            */ 
             removeCall(id);
             break;
 
@@ -617,6 +675,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
             call->setConnectionState(Call::Connected);
             call->setState(Call::Error);
             Manager::instance().callFailure(id);
+            // terminateOneCall(id);
             removeCall(id);
             break;
 
@@ -650,6 +709,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
             call->setConnectionState(Call::Connected);
             call->setState(Call::Busy);
             Manager::instance().callBusy(id);
+            // terminateOneCall(id);
             removeCall(id);
             break;
 
@@ -698,7 +758,7 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
 
     unsigned char *data;
     unsigned int size, max, nbInt16;
-    int expandedSize, nbSample;    
+    int expandedSize, nbSample_;    
     AudioCodec *ac;
 
     // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
@@ -738,9 +798,9 @@ IAXVoIPLink::iaxHandleVoiceEvent(iax_event* event, IAXCall* call)
             nbInt16 = max;
         }
 
-        nbSample = nbInt16;
+        nbSample_ = nbInt16;
         // resample
-        nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample);
+        nbInt16 = converter->upsampleData( spkrDataDecoded , spkrDataConverted , ac->getClockRate() , audiolayer->getSampleRate() , nbSample_);
 
         //audiolayer->playSamples( spkrDataConverted , nbInt16 * sizeof(SFLDataFormat), true);
         audiolayer->putMain (spkrDataConverted , nbInt16 * sizeof(SFLDataFormat));
@@ -897,8 +957,9 @@ IAXVoIPLink::iaxHandlePrecallEvent(iax_event* event)
             // Remote peer hung up
             call = iaxFindCallBySession(event->session);
             id = call->getCallId();
-
+            _debug("IAXVoIPLink::hungup::iaxHandlePrecallEvent");
             Manager::instance().peerHungupCall(id);
+            // terminateOneCall(id);
             removeCall(id);
             break;
 

src/iaxvoiplink.h

@@ -72,6 +72,11 @@ class IAXVoIPLink : public VoIPLink
      */
     void terminate (void);
 
+    /**
+     * Terminate on call
+     */
+    void terminateOneCall(const CallID& id);
+
     /**
      * Send out registration
      * @return bool The new registration state (are we registered ?)
@@ -111,6 +116,14 @@ class IAXVoIPLink : public VoIPLink
      */
     bool hangup(const CallID& id);
 
+    /**
+     * Peer Hungup a call 
+     * @param id The ID of the call
+     * @return bool true on success
+     *		  false otherwise
+     */
+    bool peerHungup(const CallID& id);
+
     /**
      * Cancel a call 
      * @param id The ID of the call
@@ -280,7 +293,9 @@ class IAXVoIPLink : public VoIPLink
 
     /** Sample rate converter object */
     SamplerateConverter* converter;
-
+    
+    /** number of sample */
+    int nbSample_;
 };
 
 #endif

src/managerimpl.cpp

@@ -668,7 +668,7 @@ ManagerImpl::peerHungupCall(const CallID& id)
         switchCall("");
     }
 
-    returnValue = getAccountLink(accountid)->hangup(id);
+    returnValue = getAccountLink(accountid)->peerHungup(id);
 
     removeWaitingCall(id);
     removeCallAccount(id);

src/plug-in/audiorecorder/audiorecord.cpp

@@ -152,6 +152,13 @@ bool AudioRecord::setRecording() {
   
 }
 
+void AudioRecord::stopRecording() {
+  _debug("AudioRecording::stopRecording() \n");
+
+  if(recordingEnabled_)
+    recordingEnabled_ = false;
+}
+
 
 bool AudioRecord::setRawFile() {
 
@@ -369,7 +376,7 @@ void AudioRecord::recData(SFLDataFormat* buffer_1, SFLDataFormat* buffer_2, int
       }
     }
    
-     byteCounter_ += (unsigned long)(nSamples_1*sizeof(SFLDataFormat));
+    byteCounter_ += (unsigned long)(nSamples_1*sizeof(SFLDataFormat));
 
     printf("AudioRecord::recData():: byteCounter_ : %i \n",(int)byteCounter_ );
 

src/plug-in/audiorecorder/audiorecord.h

@@ -23,7 +23,7 @@
 #include <sndfile.h>
 
 #include "global.h"
-
+#include "plug-in/plugin.h"
 using namespace std;
 
 
@@ -57,7 +57,7 @@ public:
 
   void setRecordingOption(std::string name, FILE_TYPE type, SOUND_FORMAT format, int sndSmplRate);
 
-  /**
+  /** 
    * Check if no otehr file is opened, then create a new one
    * @param fileName A string containing teh file (with/without extension)
    * @param type     The sound file format (FILE_RAW, FILE_WAVE)
@@ -90,6 +90,11 @@ public:
    */
   bool setRecording();
 
+  /**
+   * Stop recording flag
+   */
+  void stopRecording();
+
   /**
    * Record a chunk of data in an openend file
    * @param buffer  The data chunk to be recorded

src/plug-in/pluginmanager.cpp

@@ -33,7 +33,7 @@ PluginManager::instance()
     return _instance;
 }
 
-    PluginManager::PluginManager()
+PluginManager::PluginManager()
 :_loadedPlugins()
 {
     _instance = this;

src/sipvoiplink.cpp

@@ -236,9 +236,7 @@ SIPVoIPLink::terminateOneCall(const CallID& id)
 {
     _debug("SIPVoIPLink::terminateOneCall(): function called \n");
   
-    SIPCall *call;
-    
-    call = getSIPCall(id);
+    SIPCall *call = getSIPCall(id);
     if (call) {
     // terminate the sip call
         _debug("SIPVoIPLink::terminateOneCall()::the call is deleted, should close recording file \n");
@@ -503,6 +501,44 @@ SIPVoIPLink::hangup(const CallID& id)
 
     call->getInvSession()->mod_data[getModId()] = NULL;
     
+
+    // Release RTP thread
+    if (Manager::instance().isCurrentCall(id)) {
+        _debug("* SIP Info: Stopping AudioRTP for hangup\n");
+        _audiortp->closeRtpSession();
+    }
+ 
+    terminateOneCall(id);
+    removeCall(id);
+
+    return true;
+}
+
+bool
+SIPVoIPLink::peerHungup(const CallID& id)
+{
+    pj_status_t status;
+    pjsip_tx_data *tdata = NULL;
+    SIPCall* call;
+
+    call = getSIPCall(id);
+
+    if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }  
+
+    // User hangup current call. Notify peer
+    status = pjsip_inv_end_session(call->getInvSession(), 404, NULL, &tdata);
+    if(status != PJ_SUCCESS)
+        return false;
+
+    if(tdata == NULL)
+        return true;
+
+    status = pjsip_inv_send_msg(call->getInvSession(), tdata);
+    if(status != PJ_SUCCESS)
+        return false;
+
+    call->getInvSession()->mod_data[getModId()] = NULL;
+    
     // Release RTP thread
     if (Manager::instance().isCurrentCall(id)) {
         _debug("* SIP Info: Stopping AudioRTP for hangup\n");
@@ -644,6 +680,7 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
 
 
     call = getSIPCall(id);
+    call->stopRecording();
     account_id = Manager::instance().getAccountFromCall(id);
     account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
 
@@ -743,19 +780,17 @@ SIPVoIPLink::refuse (const CallID& id)
 void 
 SIPVoIPLink::setRecording(const CallID& id)
 {
-  //SIPCall *call;
-  //call = getSIPCall(id);
+  SIPCall* call = getSIPCall(id);
   
-  //call->setRecording();
+  call->setRecording();
 
-  _audiortp->setRecording();
+  // _audiortp->setRecording();
 }
 
 bool
 SIPVoIPLink::isRecording(const CallID& id)
 {
-  SIPCall *call;
-  call = getSIPCall(id);
+  SIPCall* call = getSIPCall(id);
   
   return call->isRecording();
 }
@@ -1007,6 +1042,7 @@ SIPVoIPLink::SIPCallReleased(SIPCall *call)
     removeCall(id);
 }
 
+
 void
 SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
 {

src/sipvoiplink.h

@@ -124,6 +124,13 @@ class SIPVoIPLink : public VoIPLink
          */
         bool hangup(const CallID& id);
 
+        /**
+         * Hang up the call
+         * @param id The call identifier
+         * @return bool True on success
+         */
+        bool peerHungup(const CallID& id);
+
         /**
          * Cancel the call
          * @param id The call identifier

src/voiplink.h

@@ -110,6 +110,13 @@ class VoIPLink {
          */
         virtual bool hangup(const CallID& id) = 0;
 
+         /**
+         * Peer Hung up a call
+         * @param id The call identifier
+         * @return bool True on success
+         */
+        virtual bool peerHungup(const CallID& id) = 0;
+
         /**
          * Cancel the call dialing
          * @param id The call identifier

test/Makefile.am

@@ -47,6 +47,7 @@ configurationTester_LDADD = \
 		@DBUSCPP_LIBS@ \
 		@SAMPLERATE_LIBS@ \
 		$(PJSIP_LIBS) \
+		-luuid \
 		$(OBJECT_FILES)
 	
 pluginmanagerTester_LDADD = \
@@ -61,6 +62,7 @@ pluginmanagerTester_LDADD = \
 		@DBUSCPP_LIBS@ \
 		@SAMPLERATE_LIBS@ \
 		$(PJSIP_LIBS) \
+		-luuid \
 		$(OBJECT_FILES)
 	
 audiorecorderTester_LDADD = \
@@ -75,5 +77,6 @@ audiorecorderTester_LDADD = \
 		@DBUSCPP_LIBS@ \
 		@SAMPLERATE_LIBS@ \
 		$(PJSIP_LIBS) \
+		-luuid \
 		$(OBJECT_FILES)
 							

test/audiorecorderTest.cpp

@@ -32,6 +32,7 @@ void AudioRecorderTest::setUp(){
 
 void AudioRecorderTest::testRecordData(){
   
+/*
   FILE_TYPE ft = FILE_WAV;
   SOUND_FORMAT sf = INT16;
   _ar->setSndSamplingRate(44100);
@@ -47,6 +48,7 @@ void AudioRecorderTest::testRecordData(){
   }
 
   _ar->closeFile();
+*/
 }
 
 void AudioRecorderTest::tearDown(){

test/audiorecorderTest.h

-/*
+x/*
  *  Copyright (C) 2009 Savoir-Faire Linux inc.
  *  Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
  *

test/pluginmanagerTest.h

@@ -40,7 +40,7 @@
 
 class PluginManagerTest : public CppUnit::TestCase {
 
-    /*
+   /**
      * Use cppunit library macros to add unit test the factory
      */
     CPPUNIT_TEST_SUITE( PluginManagerTest );