introducing namespaces

1 files changed, 16 insertions, 16 deletions

diff --git a/muse2/muse/audioconvert.cpp b/muse2/muse/audioconvert.cpp
index 95c97883..6b67f30e 100644
--- a/muse2/muse/audioconvert.cpp
+++ b/muse2/muse/audioconvert.cpp
@@ -142,7 +142,7 @@ off_t AudioConverter::readAudio(SndFileR& f, unsigned offset, float** buffer, in
   
   off_t frame     = offset;  // _spos is added before the call.
   unsigned fsrate = f.samplerate();
-  bool resample   = isValid() && ((unsigned)sampleRate != fsrate);  
+  bool resample   = isValid() && ((unsigned)MusEGlobal::sampleRate != fsrate);  
   
   // No resampling needed?
   if(!resample)
@@ -159,7 +159,7 @@ off_t AudioConverter::readAudio(SndFileR& f, unsigned offset, float** buffer, in
   {
     // Sample rates are different. Seek to a calculated 'sample rate ratio factored' position.
     
-    double srcratio = (double)fsrate / (double)sampleRate;
+    double srcratio = (double)fsrate / (double)MusEGlobal::sampleRate;
     //long inSize = long((double)frames * _src_ratio) + 1     // From MusE-2 file converter.
     off_t newfr = (off_t)floor(((double)frame * srcratio));    // From simplesynth.
   
@@ -343,13 +343,13 @@ off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n
   
 //  off_t frame     = offset;  // _spos is added before the call.
   unsigned fsrate = f.samplerate();
-  //bool resample   = src_state && ((unsigned)sampleRate != fsrate);  
-//  bool resample   = isValid() && ((unsigned)sampleRate != fsrate);  
+  //bool resample   = src_state && ((unsigned)MusEGlobal::sampleRate != fsrate);  
+//  bool resample   = isValid() && ((unsigned)MusEGlobal::sampleRate != fsrate);  
   
-  if((sampleRate == 0) || (fsrate == 0))
+  if((MusEGlobal::sampleRate == 0) || (fsrate == 0))
   {  
     #ifdef AUDIOCONVERT_DEBUG
-    printf("SRCAudioConverter::process Error: sampleRate or file samplerate is zero!\n");
+    printf("SRCAudioConverter::process Error: MusEGlobal::sampleRate or file samplerate is zero!\n");
     #endif
     return _sfCurFrame;
   }  
@@ -357,7 +357,7 @@ off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n
   SRC_DATA srcdata;
   int fchan       = f.channels();
   // Ratio is defined as output sample rate over input samplerate.
-  double srcratio = (double)sampleRate / (double)fsrate;
+  double srcratio = (double)MusEGlobal::sampleRate / (double)fsrate;
   // Extra input compensation.
   long inComp = 1;
   
@@ -369,7 +369,7 @@ off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n
   //long inSize = (long)floor(((double)outSize / srcratio));        // From simplesynth.
   //long inFrames = (long)floor(((double)outFrames / srcratio));    // From simplesynth.
   long inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp;    // From simplesynth.
-  //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(sampleRate));    // From simplesynth.
+  //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(MusEGlobal::sampleRate));    // From simplesynth.
   
   long inSize = inFrames * fchan;
   //long inSize = inFrames * channel;
@@ -583,7 +583,7 @@ RubberBandAudioConverter::RubberBandAudioConverter(int channels, int options) :
   _rbs = 0;
   _channels = channels;
   
-  _rbs = new RubberBandStretcher(sampleRate, _channels, _options);  // , initialTimeRatio = 1.0, initialPitchScale = 1.0
+  _rbs = new RubberBandStretcher(MusEGlobal::sampleRate, _channels, _options);  // , initialTimeRatio = 1.0, initialPitchScale = 1.0
 }
 
 RubberBandAudioConverter::~RubberBandAudioConverter()
@@ -605,7 +605,7 @@ void RubberBandAudioConverter::setChannels(int ch)
   _rbs = 0;
   
   _channels = ch;
-  _rbs = new RubberBandStretcher(sampleRate, _channels, _options);  // , initialTimeRatio = 1.0, initialPitchScale = 1.0
+  _rbs = new RubberBandStretcher(MusEGlobal::sampleRate, _channels, _options);  // , initialTimeRatio = 1.0, initialPitchScale = 1.0
 }
 
 void RubberBandAudioConverter::reset()
@@ -639,13 +639,13 @@ off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel
   
 //  off_t frame     = offset;  // _spos is added before the call.
   unsigned fsrate = f.samplerate();
-  //bool resample   = src_state && ((unsigned)sampleRate != fsrate);  
-//  bool resample   = isValid() && ((unsigned)sampleRate != fsrate);  
+  //bool resample   = src_state && ((unsigned)MusEGlobal::sampleRate != fsrate);  
+//  bool resample   = isValid() && ((unsigned)MusEGlobal::sampleRate != fsrate);  
   
-  if((sampleRate == 0) || (fsrate == 0))
+  if((MusEGlobal::sampleRate == 0) || (fsrate == 0))
   {  
     #ifdef AUDIOCONVERT_DEBUG
-    printf("RubberBandAudioConverter::process Error: sampleRate or file samplerate is zero!\n");
+    printf("RubberBandAudioConverter::process Error: MusEGlobal::sampleRate or file samplerate is zero!\n");
     #endif
     return _sfCurFrame;
   }  
@@ -653,7 +653,7 @@ off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel
 //  SRC_DATA srcdata;
   int fchan       = f.channels();
   // Ratio is defined as output sample rate over input samplerate.
-  double srcratio = (double)sampleRate / (double)fsrate;
+  double srcratio = (double)MusEGlobal::sampleRate / (double)fsrate;
   // Extra input compensation.
   long inComp = 1;
   
@@ -665,7 +665,7 @@ off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel
   //long inSize = (long)floor(((double)outSize / srcratio));        // From simplesynth.
   //long inFrames = (long)floor(((double)outFrames / srcratio));    // From simplesynth.
   long inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp;    // From simplesynth.
-  //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(sampleRate));    // From simplesynth.
+  //long inFrames = (long)floor(double(outFrames * sfinfo.samplerate) / double(MusEGlobal::sampleRate));    // From simplesynth.
   
   long inSize = inFrames * fchan;
   //long inSize = inFrames * channel;