Merge branch 'debug_threads'

Conflicts:

	configure.ac
	src/Makefile.am
	src/audio/alsalayer.cpp
	src/audio/audiortp.cpp

autogen.sh

@@ -4,6 +4,6 @@
 aclocal -I m4
 libtoolize --force
 autoheader
-autoconf -f
+autoconf -v -f
 automake -a
 ./configure $@

configure.ac

@@ -36,7 +36,10 @@ AC_CONFIG_FILES([src/Makefile \
   src/audio/codecs/Makefile
   src/audio/codecs/ilbc/Makefile \
   src/config/Makefile \
-  src/dbus/Makefile ]) 
+  src/dbus/Makefile \
+  src/plug-in/audiorecorder/Makefile \
+  src/plug-in/Makefile \
+  src/plug-in/test/Makefile]) 
   
   dnl Unitary test section
 AC_CONFIG_FILES([test/Makefile])

globals.mak

 # Global variables
 src=$(top_srcdir)
 sflcodecdir=$(libdir)/sflphone/codecs
+sflplugindir=$(libdir)/sflphone/plugins
 
 PJSIP_LIBS = -lpjnath-sfl -lpjsua-sfl -lpjsip-sfl -lpjmedia-sfl -lpjsip-simple-sfl -lpjsip-ua-sfl -lpjmedia-codec-sfl -lpjlib-util-sfl -lpj-sfl 
 
@@ -16,5 +17,6 @@ AM_CPPFLAGS = \
 	@SIP_CFLAGS@ \
 	@DBUSCPP_CFLAGS@ \
 	-DCODECS_DIR=\""$(sflcodecdir)"\" \
+	-DPLUGINS_DIR=\""$(sflplugindir)"\" \
 	-DENABLE_TRACE
 

src/Makefile.am

@@ -13,7 +13,7 @@ IAXSOURCES =
 IAXHEADERS =
 endif
 
-SUBDIRS = audio config dbus
+SUBDIRS = audio config dbus plug-in
 
 # Add here the cpp files to be build with sflphone
 sflphoned_SOURCES = \
@@ -75,6 +75,8 @@ libsflphone_la_LIBADD = \
 	./audio/libaudio.la \
 	./dbus/libdbus.la \
 	./config/libconfig.la \
+	./plug-in/libplugin.la \
+	./plug-in/audiorecorder/libaudiorecorder.la \
 	$(IAX_LIBS) 
 
 libsflphone_la_SOURCES =

src/account.h

@@ -96,6 +96,8 @@ class Account{
          */
         inline VoIPLink* getVoIPLink() { return _link; }
 
+        inline void setVoIPLink (VoIPLink *link) { _link = link; }
+
         /**
          * Register the underlying VoIPLink. Launch the event listener.
          * This should update the getRegistrationState() return value.

src/audio/alsalayer.cpp

@@ -193,9 +193,10 @@ void AlsaLayer::startCaptureStream (void)
 void AlsaLayer::prepareCaptureStream (void)
 {
     if (is_capture_open() ) {
-        _debug("Prepare the capture\n");
-        if(snd_pcm_prepare (_CaptureHandle) < 0)    _debug("Error preparing the device\n");
-        prepare_capture ();
+        if(snd_pcm_prepare (_CaptureHandle) < 0)    
+            _debug("");
+        else
+            prepare_capture ();
     }
 }
 
@@ -383,11 +384,11 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
     }
 
     /* Start the secondary audio thread for callbacks */
-    try {
+    try{
         _audioThread->start();
     }
-    catch(...){
-        _debug("Fail to start audio thread\n");
+    catch (...) {
+        _debugException("Fail to start audio thread\n");
     }
 
     return true;

src/audio/audiorecord.cpp

+/*
+ *  Copyright (C) 2008 Savoir-Faire Linux inc.
+ *  Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 3 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include "audiorecord.h"
+
+AudioRecord::AudioRecord(){
+  sndSmplRate_ = 44100;
+  channels_ = 1;
+  byteCounter_ = 0;
+
+}
+
+
+void AudioRecord::setSndSamplingRate(int smplRate){
+  sndSmplRate_ = smplRate;
+}
+
+
+void AudioRecord::openFile(std::string fileName, FILE_TYPE type, SOUND_FORMAT format) {
+
+  channels_ =1;
+  fileType_ = type;
+  byteCounter_ = 0;
+  sndFormat_ = format;
+
+  bool result = false;
+
+  if(fileType_ == FILE_RAW){
+    result = setRawFile( fileName.c_str() );
+  }
+  else if (fileType_ == FILE_WAV){
+    result = setWavFile( fileName.c_str() );
+  }
+
+}
+
+
+void AudioRecord::closeFile() {
+
+  if (fp == 0) return;
+
+  if (fileType_ == FILE_RAW)
+    fclose(fp);
+  else if (fileType_ == FILE_WAV)
+    this->closeWavFile();
+
+}
+
+
+bool AudioRecord::isOpenFile() {
+  
+  if(fp)
+    return true;
+  else
+    return false;
+}
+
+
+bool AudioRecord::setRawFile(const char *fileName) {
+
+  char name[8192];
+  strncpy(name, fileName, 8192);
+  if ( strstr(name, ".raw") == NULL) strcat(name, ".raw");
+  fp = fopen(name, "wb");
+  if ( !fp ) {
+    cout << "AudioRecord: could not create RAW file: " << name << '.';
+    return false;
+  }
+
+  if ( sndFormat_ != INT16 ) { // TODO need to change INT16 to SINT16
+    sndFormat_ = INT16;
+    cout << "AudioRecord: using 16-bit signed integer data format for file " << name << '.';
+  }
+
+  cout << "AudioRecord: creating RAW file: " << name;
+  return true;
+}
+
+
+bool AudioRecord::setWavFile(const char *fileName) {
+  
+  char name[8192];
+  strncpy(name, fileName, 8192);
+  if ( strstr(name, ".wav") == NULL) strcat(name, ".wav");
+  fp = fopen(name, "wb");
+  if ( !fp ) {
+    cout << "AudioRecord: could not create WAV file: " << name;
+    return false;
+  }
+
+  struct wavhdr hdr = {"RIF", 44, "WAV", "fmt", 16, 1, 1,
+                        44100, 0, 2, 16, "dat", 0};
+  hdr.riff[3] = 'F';
+  hdr.wave[3] = 'E';
+  hdr.fmt[3]  = ' ';
+  hdr.data[3] = 'a';
+  hdr.num_chans = channels_;
+  if ( sndFormat_ == INT16 ) { //  TODO need to write INT16 to SINT16
+    hdr.bits_per_samp = 16;
+  }
+  hdr.bytes_per_samp = (SINT16) (channels_ * hdr.bits_per_samp / 8);
+  hdr.bytes_per_sec = (SINT32) (hdr.sample_rate * hdr.bytes_per_samp);
+
+
+  if ( fwrite(&hdr, 4, 11, fp) != 11 ) {
+    cout << "AudioRecord: could not write WAV header for file " << name << '.';
+    return false;
+  }
+
+  cout << "AudioRecord: creating WAV file: " << name;
+  return true;
+}
+
+
+void AudioRecord::closeWavFile()
+{
+  int bytes_per_sample = 1;
+  if ( sndFormat_ == INT16 )
+    bytes_per_sample = 2;
+
+
+  SINT32 bytes = byteCounter_ * channels_ * bytes_per_sample;
+  fseek(fp, 40, SEEK_SET); // jump to data length
+  fwrite(&bytes, 4, 1, fp);
+
+  bytes = byteCounter_ * channels_ * bytes_per_sample + 44; // + 44 for the wave header
+  fseek(fp, 4, SEEK_SET); // jump to file size
+  fwrite(&bytes, 4, 1, fp);
+  fclose( fp );
+}
+
+
+void AudioRecord::recData(SFLDataFormat* buffer, int nSamples) {
+
+  if (fp == 0){
+    cout << "AudioRecord: Can't record data, a file has not yet been opened!";
+    return;
+  }
+ 
+  if ( sndFormat_ == INT16 ) { // TODO change INT16 to SINT16
+    if (nSamples <= 1){
+      if ( fwrite(buffer, 2, 1, fp) != 1)
+        cout << "AudioRecord: Could not record data!";
+    }
+    else { 
+      for ( int k=0; k<nSamples; k++ ) {
+        cout << "Buffer[" << k << "] : " << buffer[k] << "\n";
+        if ( fwrite(&buffer[k], 2, 1, fp) != 1 )
+          cout << "AudioRecord: Could not record data!";
+      }
+    }
+  }
+
+  
+
+
+  byteCounter_ += (unsigned long)(sizeof(buffer) / sizeof(SINT16));
+
+  return;
+}
+

src/audio/audiorecord.h

+/*
+ *  Copyright (C) 2008 Savoir-Faire Linux inc.
+ *  Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 3 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+
+#include <iostream>
+#include <string.h>
+
+#include "global.h"
+
+using namespace std;
+
+
+// structure for the wave header
+struct wavhdr {
+  char riff[4];           // "RIFF"
+  SINT32 file_size;       // in bytes
+  char wave[4];           // "WAVE"
+  char fmt[4];            // "fmt "
+  SINT32 chunk_size;      // in bytes (16 for PCM)
+  SINT16 format_tag;      // 1=PCM, 2=ADPCM, 3=IEEE float, 6=A-Law, 7=Mu-Law
+  SINT16 num_chans;       // 1=mono, 2=stereo
+  SINT32 sample_rate;
+  SINT32 bytes_per_sec;
+  SINT16 bytes_per_samp;  // 2=16-bit mono, 4=16-bit stereo
+  SINT16 bits_per_samp;
+  char data[4];           // "data"
+  SINT32 data_length;     // in bytes
+};
+
+class AudioRecord
+{
+
+public:
+
+  AudioRecord();
+
+  void setSndSamplingRate(int smplRate);
+
+  /**
+   * 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)
+   * @param format   Internal sound format (INT16 / INT32)
+   */
+  void openFile(std::string fileName, FILE_TYPE type, SOUND_FORMAT format);
+
+  /**
+   * Close the opend recording file. If wave: cout the number of byte
+   */
+  void closeFile();
+
+  /**
+   * Check if a file is already opened
+   */
+  bool isOpenFile();
+
+  /**
+   * Record a chunk of data in an openend file
+   * @param buffer  The data chunk to be recorded
+   * @param nSamples Number of samples (number of bytes) to be recorded 
+   */
+  void recData(SFLDataFormat* buffer, int nSamples); // TODO ad the data to rec
+
+protected:
+
+  /**
+   * Set the header for raw files
+   */
+  bool setRawFile(const char* fileName);
+
+  /**
+   * Set the header for wave files
+   */
+  bool setWavFile(const char* fileName);
+
+  /**
+   * Compute the number of byte recorded and close the file
+   */
+  void closeWavFile();
+
+  /**
+   * Pointer to the recorded file
+   */
+  FILE *fp;                      //file pointer
+
+  /**
+   * File format (RAW / WAVE)
+   */
+  FILE_TYPE fileType_;
+
+  /**
+   * Sound format (SINT16/SINT32)
+   */
+  SOUND_FORMAT sndFormat_;
+
+  /**
+   * Number of channels
+   */
+  int channels_;
+
+  /**
+   * Number f byte recorded
+   */
+  unsigned long byteCounter_;
+
+  /**
+   * Sampling rate
+   */
+  int sndSmplRate_;
+
+};

src/audio/audiortp.cpp

@@ -82,7 +82,7 @@ void
 AudioRtp::closeRtpSession () {
   ost::MutexLock m(_threadMutex);
   // This will make RTP threads finish.
-  // _debug("Stopping AudioRTP\n");
+  _debug("Stopping AudioRTP\n");
   try {
     delete _RTXThread; _RTXThread = 0;
   } catch(...) {
@@ -126,7 +126,6 @@ AudioRtpRTX::~AudioRtpRTX () {
     _debugException("! ARTP: Thread destructor didn't terminate correctly");
     throw;
   }
-  //_debug("terminate audiortprtx ended...\n");
   _ca = 0;
   if (!_sym) {
     delete _sessionRecv; _sessionRecv = NULL;
@@ -145,6 +144,7 @@ AudioRtpRTX::~AudioRtpRTX () {
   delete time; time = NULL;
 
   delete converter; converter = NULL;
+  
 }
 
   void

src/audio/pulselayer.cpp

@@ -44,16 +44,10 @@ static  void audioCallback ( pa_stream* s, size_t bytes, void* userdata )
 // Destructor
 PulseLayer::~PulseLayer (void) 
 { 
-    //closeLayer();
-  
-    /* Delete the pointer streams */
-    delete playback;
-    delete record;
-  
+    closeLayer ();
+
     pa_context_disconnect( context );  
     pa_context_unref( context );
-
-    sleep(2);
 }
 
   void

src/audio/pulselayer.h

@@ -22,6 +22,7 @@
 
 #include "audiolayer.h"
 #include "audiostream.h"
+#include "plug-in/audiorecorder/audiorecord.h"
 
 #include <pulse/pulseaudio.h>
 

src/eventthread.cpp

@@ -54,6 +54,7 @@ void AudioThread::run (void)
 {
     while(!testCancel()) {
         _alsa->audioCallback();
+        Thread::sleep(3);
     }
 }
 

src/global.h

@@ -37,6 +37,19 @@
 typedef float float32;
 typedef short int16;
 
+//useful typedefs.
+typedef signed short SINT16;
+typedef signed int SINT32;
+
+typedef unsigned long FILE_TYPE;
+typedef unsigned long SOUND_FORMAT;
+
+const FILE_TYPE  FILE_RAW = 1;
+const FILE_TYPE  FILE_WAV = 2;
+
+static const SOUND_FORMAT INT16 = 0x2; // TODO shold change these symbols
+static const SOUND_FORMAT INT32 = 0x8;
+
 #define SUCCESS                 0
 
 #define ASSERT( expected , value)       if( value == expected ) return SUCCESS; \

src/managerimpl.cpp

@@ -249,6 +249,7 @@ ManagerImpl::answerCall(const CallID& id)
   bool
 ManagerImpl::hangupCall(const CallID& id)
 {
+  _debug("ManagerImpl::hangupCall(): This function is called when user hangup \n");
     PulseLayer *pulselayer;
     AccountID accountid;
     bool returnValue;
@@ -370,6 +371,7 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
   bool
 ManagerImpl::refuseCall (const CallID& id)
 {
+  _debug("ManagerImpl::refuseCall(): method called");
   stopTone(true);
   AccountID accountid = getAccountFromCall( id );
   if (accountid == AccountNULL) {
@@ -510,6 +512,7 @@ ManagerImpl::playDtmf(char code, bool isTalking)
         // Put buffer to urgentRingBuffer 
         // put the size in bytes...
         // so size * 1 channel (mono) * sizeof (bytes for the data)
+        audiolayer->startStream();
         audiolayer->putUrgent (buf, size * sizeof(SFLDataFormat));
     }
     ret = true;
@@ -631,8 +634,10 @@ ManagerImpl::peerRingingCall(const CallID& id)
   void
 ManagerImpl::peerHungupCall(const CallID& id)
 {
+    _debug("ManagerImpl::peerHungupCall():this function is called when peer hangup \n");
     PulseLayer *pulselayer;
     AccountID accountid;
+    bool returnValue;
 
     accountid = getAccountFromCall( id );
     if (accountid == AccountNULL) {
@@ -648,6 +653,8 @@ ManagerImpl::peerHungupCall(const CallID& id)
         switchCall("");
     }
 
+    returnValue = getAccountLink(accountid)->hangup(id);
+
     removeWaitingCall(id);
     removeCallAccount(id);
   
@@ -1408,9 +1415,14 @@ int ManagerImpl::isStunEnabled (void)
 
 void ManagerImpl::enableStun (void)
 {
-  ( getConfigInt( SIGNALISATION , STUN_ENABLE ) == STUN_ENABLED )? setConfig(SIGNALISATION , STUN_ENABLE , NO_STR ) : setConfig( SIGNALISATION , STUN_ENABLE , YES_STR );
+    /* Update the config */
+    ( getConfigInt( SIGNALISATION , STUN_ENABLE ) == STUN_ENABLED )? setConfig(SIGNALISATION , STUN_ENABLE , NO_STR ) : setConfig( SIGNALISATION , STUN_ENABLE , YES_STR );
+
+    /* Restart PJSIP */
+    this->restartPJSIP (); 
 }
 
+
     int
 ManagerImpl::getVolumeControls( void )
 {
@@ -1945,6 +1957,7 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
     setConfig(accountID, CONFIG_ACCOUNT_MAILBOX,(*details.find(CONFIG_ACCOUNT_MAILBOX)).second);
   
     // SIP SPECIFIC
+    /*
     if (accountType == "SIP") {
     
         link =  Manager::instance().getAccountLink( accountID );
@@ -1966,8 +1979,8 @@ void ManagerImpl::setAccountDetails( const std::string& accountID, const std::ma
         {
             link->setStunServer("");
         }
-        //restartPjsip();
-    }
+        restartPJSIP();
+    }*/
 
     saveConfig();
   
@@ -2232,6 +2245,30 @@ AccountMap ManagerImpl::getSipAccountMap( void )
     return sipaccounts;
 }
 
+void ManagerImpl::restartPJSIP (void)
+{
+    //SIPVoIPLink *siplink;
+
+    //unloadAccountMap ();
+
+    /* First unregister all SIP accounts */
+    //this->unregisterCurSIPAccounts();
+    /* Terminate and initialize the PJSIP library */
+    //siplink = dynamic_cast<SIPVoIPLink*> (getSIPAccountLink ());
+    //if (siplink) 
+    //{
+      //  siplink->terminate ();
+       // _debug ("*************************************************Terminate done\n");
+        //siplink = SIPVoIPLink::instance("");
+        //siplink->init ();
+    //}
+    //_debug("***************************************************Init Done\n");
+    //loadAccountMap();
+    //initRegisterAccounts ();
+    /* Then register all enabled SIP accounts */
+    //this->registerCurSIPAccounts(siplink);
+}
+
 VoIPLink* ManagerImpl::getAccountLink(const AccountID& accountID)
 {
   Account* acc = getAccount(accountID);
@@ -2241,6 +2278,23 @@ VoIPLink* ManagerImpl::getAccountLink(const AccountID& accountID)
   return 0;
 }
 
+VoIPLink* ManagerImpl::getSIPAccountLink()
+{
+    /* We are looking for the first SIP account we met because all the SIP accounts have the same voiplink */
+    Account *account;
+    AccountMap::iterator iter;
+    for(iter = _accountMap.begin(); iter != _accountMap.end(); ++iter) {
+        account = iter->second;
+        if( account->getType() == "sip" ){
+            return account->getVoIPLink();
+        }
+    }
+    return NULL;
+}
+
+
+
+
 pjsip_regc 
 *getSipRegcFromID(const AccountID& id UNUSED)
 {
@@ -2253,32 +2307,35 @@ pjsip_regc
 
 void ManagerImpl::unregisterCurSIPAccounts()
 {
-  AccountMap::iterator iter = _accountMap.begin();
-  while( iter != _accountMap.end() ) {
-    if ( iter->second) {
-        std::string p =  Manager::instance().getConfigString( iter->first , CONFIG_ACCOUNT_TYPE );
-      if ( iter->second->isEnabled() && p == "SIP") {
-	// NOW
-	iter->second->unregisterVoIPLink();
-      }
-    }
+    Account *current;
+
+    AccountMap::iterator iter = _accountMap.begin();
+    while( iter != _accountMap.end() ) {
+        current = iter->second;
+        if (current) {
+            if ( current->isEnabled() && current->getType() == "sip") {
+	            current->unregisterVoIPLink();