Commit a6654198 authored by Emmanuel Milou's avatar Emmanuel Milou

Merge branch 'recording'

parents 734bc33a 0e429104
......@@ -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));
......
......@@ -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);
}
......
......@@ -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;
};
......
......@@ -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];
......
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