Merge branch 'recording'

sflphone-gtk/src/calltree.c

@@ -691,7 +691,6 @@ update_call_tree (calltab_t* tab, call_t * c)
 	}
 	else
 	{
-          g_print("Stuff to be printed  %s  %s \n",call_get_number(c),call_get_name(c));
 	  description = g_markup_printf_escaped("<b>%s</b> <i>%s</i>", 
 	      call_get_number(c),
 	      call_get_name(c)); 

sflphone-gtk/src/dbus.c

@@ -66,7 +66,7 @@ curent_selected_codec (DBusGProxy *proxy UNUSED,
                   const gchar* codecName,
                   void * foo  UNUSED )
 {
-  g_print ("Codec decided! %s\n",codecName);
+  g_print ("%s codec decided for call %s\n",codecName,callID);
   sflphone_display_selected_codec (codecName);
 }
 

src/plug-in/audiorecorder/audiorecord.h

@@ -26,7 +26,7 @@
 #include <sstream>
 
 #include "plug-in/plugin.h"
-#include "audiodsp.h"
+// #include "audiodsp.h"
 
 // class AudioDSP;
 
@@ -189,11 +189,6 @@ protected:
   std::string savePath_;
 
   std::string call_id_;
-  
-  /**
-   * AudioDSP test (compute RMS value)
-   */
-  AudioDSP dsp;
  
 };
 

src/samplerateconverter.cpp

@@ -64,8 +64,8 @@ void SamplerateConverter::init( void ) {
   // libSamplerateConverter-related
   // Set the converter type for the upsampling and the downsampling
   // interpolator SRC_SINC_BEST_QUALITY
-  _src_state_mic  = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
-  _src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
+  _src_state_mic  = src_new(SRC_LINEAR, 1, &_src_err);
+  _src_state_spkr = src_new(SRC_LINEAR, 1, &_src_err);
 
   int nbSamplesMax = (int) ( getFrequence() * getFramesize() / 1000 );
   _floatBufferDownMic  = new float32[nbSamplesMax];