Debuging messages

src/audio/alsalayer.cpp

@@ -122,7 +122,7 @@ AlsaLayer::startStream(void)
     void
 AlsaLayer::stopStream(void) 
 {
-    _debug ("Stop ALSA streams\n");
+    _debug ("AlsaLayer::stopStream :: Stop ALSA streams\n");
     stopCaptureStream ();
     //stopPlaybackStream ();
 
@@ -190,9 +190,13 @@ bool AlsaLayer::isCaptureActive(void) {
     
 void AlsaLayer::stopCaptureStream (void)
 {
+    int err;  
+
     if(_CaptureHandle){
-        snd_pcm_drop (_CaptureHandle);
+        err = snd_pcm_drop (_CaptureHandle);
+        _debug("AlsaLayer::stopCaptureStream snd_pcm_drop returned vaue : %i\n",err);
         stop_capture ();
+        _debug("Wroking here !!!!!!!!!!!!!!!\n");
     }
 }
 
@@ -373,7 +377,8 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
     
     if(flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK)
     {
-        if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(),  SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
+      // if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(),  SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
+        if((err = snd_pcm_open(&_PlaybackHandle, PCM_PLUGHW,  SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
             _debugAlsa("Error while opening playback device %s\n",  pcm_p.c_str());
             setErrorMessage( ALSA_PLAYBACK_DEVICE );
             close_playback ();

src/audio/pulselayer.cpp

@@ -248,7 +248,8 @@ PulseLayer::processData( void )
         // If the playback buffer is full, we don't overflow it; wait for it to have free space
         if( pa_stream_writable_size(playback->pulseStream()) == 0 )
             return;
-
+        
+        _debug("PulseLayer::processData() \n");
         writeToSpeaker();
     }
 }
@@ -256,6 +257,7 @@ PulseLayer::processData( void )
     void
 PulseLayer::writeToSpeaker( void )
 {   
+     _debug("PulseLayer::writeToSpeaker() \n");
     /** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
     int urgentAvail; 
     /** Bytes available in the regular ringbuffer ( reserved for voice ) */

src/audio/tone.cpp

@@ -114,6 +114,8 @@ Tone::genBuffer(const std::string& definition)
 void
 Tone::genSin(SFLDataFormat* buffer, int frequency1, int frequency2, int nb) 
 {
+
+  _debug("Tone::genSin :: called!!!!!!!!!!!!!!!!!! %i \n",nb);
   double pi2 = 6.28318520; 
   double var1 = pi2 * (double)frequency1 / (double)_sampleRate;
   double var2 = pi2 * (double)frequency2 / (double)_sampleRate;

src/audio/tonegenerator.cpp

@@ -45,7 +45,9 @@ ToneGenerator::~ToneGenerator (void) {
 void
 ToneGenerator::generateSin (int lowerfreq, int higherfreq, int16* ptr, int len) const {
   double var1, var2;
-													
+			
+  _debug("when is this called"); 
+ 										
   var1 = (double)2 * (double)M_PI * (double)higherfreq / (double)_sampleRate; 
   var2 = (double)2 * (double)M_PI * (double)lowerfreq / (double)_sampleRate;
 

src/managerimpl.cpp

@@ -313,6 +313,7 @@ ManagerImpl::cancelCall (const CallID& id)
   bool
 ManagerImpl::onHoldCall(const CallID& id)
 {
+  _debug("*************** ON HOLD ***********************************\n");
   stopTone(true);
   AccountID accountid = getAccountFromCall( id );
   if (accountid == AccountNULL) {
@@ -335,6 +336,7 @@ ManagerImpl::onHoldCall(const CallID& id)
   bool
 ManagerImpl::offHoldCall(const CallID& id)
 {
+  _debug("*************** OFF HOLD ***********************************\n");
   stopTone(false);
   AccountID accountid = getAccountFromCall( id );
   if (accountid == AccountNULL) {

src/plug-in/audiorecorder/Makefile.am

@@ -3,6 +3,4 @@ include $(top_srcdir)/globals.mak
 noinst_LTLIBRARIES = libaudiorecorder.la
 
 libaudiorecorder_la_SOURCES = \
-		audiorecord.cpp \
-		audiodsp.h \
-		audiodsp.cpp
\ No newline at end of file
+		audiorecord.cpp
\ No newline at end of file

src/sipvoiplink.cpp

@@ -805,15 +805,7 @@ std::string
 SIPVoIPLink::getCurrentCodecName()
 {
 
-  _debug("SIPVoIPLink::getCurrentCodecName Ok 1 : \n");
- 
-  printf("AAAAAAAAAAAAAHHHHHHH!!!!!!:: %s \n",Manager::instance().getCurrentCallId().c_str());
-
-  SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());
-  
-  _debug("SIPVoIPLink::getCurrentCodecName Ok 2 : \n");
-  
-  printf("OOOOOOOUUUUUUPPPPPPPPPPSSSSSSSSSSS!!!!!!:: %s \n",call->getCallId().c_str());  
+  SIPCall *call = getSIPCall(Manager::instance().getCurrentCallId());  
 
   AudioCodec *ac = call->getCodecMap().getCodec(call->getAudioCodec());