new branch muse2, first checkin

1 files changed, 886 insertions, 0 deletions

diff --git a/muse2/muse/audioconvert.cpp b/muse2/muse/audioconvert.cpp
new file mode 100644
index 00000000..878bc8e7
--- /dev/null
+++ b/muse2/muse/audioconvert.cpp
@@ -0,0 +1,886 @@
+//=========================================================
+//  MusE
+//  Linux Music Editor
+//  $Id: audioconvert.cpp,v 1.1.1.1 2009/12/28 16:07:33 terminator356 Exp $
+//
+//  (C) Copyright 1999-2009 Werner Schweer (ws@seh.de)
+//
+//  Audio converter module created by Tim terminator356
+//=========================================================
+
+#include <math.h>
+
+#include "wave.h"
+#include "globals.h"
+#include "audioconvert.h"
+#include "eventbase.h"
+
+//#define AUDIOCONVERT_DEBUG
+//#define AUDIOCONVERT_DEBUG_PRC
+
+//---------------------------------------------------------
+//   AudioConvertMap
+//---------------------------------------------------------
+
+void AudioConvertMap::remapEvents(const EventList* el)  
+{
+
+}
+
+iAudioConvertMap AudioConvertMap::addEvent(EventBase* eb)
+{
+  iAudioConvertMap iacm = getConverter(eb);
+  if(iacm == end())
+  {
+    AudioConverter* cv = 0;
+    if(!eb->sndFile().isNull())
+      cv = new SRCAudioConverter(eb->sndFile().channels(), SRC_SINC_MEDIUM_QUALITY);
+    
+    // Use insert with hint for speed.
+    return insert(iacm, std::pair<EventBase*, AudioConverter*> (eb, cv));
+  }
+  else
+    // Adopt a policy of returning an already existing item to enforce no-duplicates.
+    return iacm;
+}
+
+void AudioConvertMap::removeEvent(EventBase* eb)
+{
+  iAudioConvertMap iacm = find(eb);
+  if(iacm != end())
+  {
+    AudioConverter* cv = iacm->second;
+    if(cv)
+      delete cv;
+    erase(iacm);  
+  }    
+}
+
+iAudioConvertMap AudioConvertMap::getConverter(EventBase* eb)
+{
+  return find(eb);
+}
+
+//---------------------------------------------------------
+//   AudioConverter
+//---------------------------------------------------------
+
+AudioConverter::AudioConverter()
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("AudioConverter::AudioConverter this:%p\n", this);
+  #endif
+
+  _refCount = 1;
+  _sfCurFrame = 0;
+}
+
+AudioConverter::~AudioConverter()
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("AudioConverter::~AudioConverter this:%p\n", this);
+  #endif
+}
+
+AudioConverter* AudioConverter::reference()
+{
+  _refCount += 1;
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("AudioConverter::reference this:%p current refcount:%d\n", this, _refCount);
+  #endif
+  return this;
+}
+
+AudioConverter* AudioConverter::release(AudioConverter* cv)
+{
+  if(!cv)
+    return 0;
+  //if(cv->incRefCount(-1) <= 0)
+  cv->_refCount -= 1;
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("AudioConverter::release converter:%p current refcount:%d\n", cv, cv->_refCount);
+  #endif
+  if(cv->_refCount <= 0)
+  {
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("AudioConverter::release deleting converter:%p\n", cv);
+    #endif
+    delete cv;
+    cv = 0;
+  }
+  return cv;  
+}
+
+//off_t AudioConverter::readAudio(SndFileR& f, off_t sfCurFrame, unsigned offset, float** buffer, int channel, int n, bool doSeek, bool overwrite)
+off_t AudioConverter::readAudio(SndFileR& f, unsigned offset, float** buffer, int channel, int n, bool doSeek, bool overwrite)
+{
+  if(f.isNull())
+    return _sfCurFrame;
+  
+  // Added by Tim. p3.3.17
+  //#ifdef AUDIOCONVERT_DEBUG_PRC
+  //printf("AudioConverter::process %s audConv:%p sfCurFrame:%ld offset:%u channel:%d fchan:%d n:%d\n", 
+  //        f.name().latin1(), this, sfCurFrame, offset, channel, f.channels(), n);
+  //#endif
+  
+  off_t frame     = offset;  // _spos is added before the call.
+  unsigned fsrate = f.samplerate();
+  bool resample   = isValid() && ((unsigned)sampleRate != fsrate);  
+  
+  // No resampling needed?
+  if(!resample)
+  {
+    // Sample rates are the same. Just a regular seek + read, no conversion.
+    _sfCurFrame = f.seek(frame, 0);
+    return _sfCurFrame + f.read(channel, buffer, n, overwrite);
+  }
+  
+  // Is a 'transport' seek requested? (Not to be requested with every read! Should only be for 'first read' seeks, or positional 'transport' seeks.)
+  // Due to the support of sound file references in MusE, seek must ALWAYS be done before read, as before,
+  //  except now we alter the seek position if sample rate conversion is being used and remember the seek positions. 
+  if(doSeek)
+  {
+    // Sample rates are different. Seek to a calculated 'sample rate ratio factored' position.
+    
+    double srcratio = (double)fsrate / (double)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.
+  
+    _sfCurFrame = f.seek(newfr, 0);
+    
+    // Added by Tim. p3.3.17
+    //#ifdef AUDIOCONVERT_DEBUG_PRC
+    //printf("AudioConverter::process Seek frame:%ld converted to frame:%ld sfCurFrame:%ld\n", frame, newfr, sfCurFrame);
+    //#endif
+    
+    // Reset the converter. Its current state is meaningless now.
+    reset();
+  }
+  else  
+  {
+    // No seek requested. 
+    // Added by Tim. p3.3.17
+    //#ifdef AUDIOCONVERT_DEBUG_PRC
+    //printf("AudioConverter::process No 'transport' seek, rates different. Seeking to sfCurFrame:%ld\n", sfCurFrame);
+    //#endif
+    
+    // Sample rates are different. We can't just tell seek to go to an absolute calculated position, 
+    //  since the last position can vary - it might not be what the calculated position is. 
+    // We must use the last position left by SRC conversion, ie. let the file position progress on its own.
+    _sfCurFrame = f.seek(_sfCurFrame, 0);
+  }
+  
+  /*
+  int   fchan              = f.channels();
+  long  outFrames          = n;  
+  long  outSize            = outFrames * fchan;
+  float outbuffer[outSize];
+  */
+  
+  //sfCurFrame = process(f, sfCurFrame, offset, &outbuffer[0], channel, n);
+//  sfCurFrame = process(f, sfCurFrame, outbuffer, channel, n);
+  //sfCurFrame = process(f, sfCurFrame, buffer, channel, n, overwrite);
+  _sfCurFrame = process(f, buffer, channel, n, overwrite);
+  
+  /*
+  float*  poutbuf = &outbuffer[0];
+  if(fchan == channel) 
+  {
+    if(overwrite)
+      //for (size_t i = 0; i < rn; ++i) 
+      for (int i = 0; i < n; ++i) 
+      {
+        for(int ch = 0; ch < channel; ++ch)
+          *(buffer[ch] + i) = *poutbuf++;
+      }
+    else
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        for(int ch = 0; ch < channel; ++ch)
+          *(buffer[ch] + i) += *poutbuf++;
+      }
+  }
+  else if((fchan == 2) && (channel == 1)) 
+  {
+    // stereo to mono
+    if(overwrite)
+      //for(size_t i = 0; i < rn; ++i)
+      for(int i = 0; i < n; ++i)
+        *(buffer[0] + i) = poutbuf[i + i] + poutbuf[i + i + 1];
+    else  
+      //for(size_t i = 0; i < rn; ++i)
+      for(int i = 0; i < n; ++i)
+        *(buffer[0] + i) += poutbuf[i + i] + poutbuf[i + i + 1];
+  }
+  else if((fchan == 1) && (channel == 2)) 
+  {
+    // mono to stereo
+    if(overwrite)
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        float data = *poutbuf++;
+        *(buffer[0]+i) = data;
+        *(buffer[1]+i) = data;
+      }
+    else  
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        float data = *poutbuf++;
+        *(buffer[0]+i) += data;
+        *(buffer[1]+i) += data;
+      }
+  }
+  else 
+  {
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("AudioConverter::readAudio Channel mismatch: source chans:%d -> dst chans:%d\n", fchan, channel);
+    #endif
+  }
+  */
+  
+  return _sfCurFrame;
+}
+
+//---------------------------------------------------------
+//   SRCAudioConverter
+//---------------------------------------------------------
+
+SRCAudioConverter::SRCAudioConverter(int channels, int type) : AudioConverter()
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("SRCAudioConverter::SRCAudioConverter this:%p channels:%d type:%d\n", this, channels, type);
+  #endif
+
+  _type = type;
+  _src_state = 0;
+  _channels = channels;
+  
+  int srcerr;
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("SRCAudioConverter::SRCaudioConverter Creating samplerate converter type:%d with %d channels\n", _type, _channels);
+  #endif
+  _src_state = src_new(_type, _channels, &srcerr); 
+  if(!_src_state)      
+    printf("SRCAudioConverter::SRCaudioConverter Creation of samplerate converter type:%d with %d channels failed:%s\n", _type, _channels, src_strerror(srcerr));
+}
+
+SRCAudioConverter::~SRCAudioConverter()
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("SRCAudioConverter::~SRCAudioConverter this:%p\n", this);
+  #endif
+  if(_src_state)
+    src_delete(_src_state);
+}
+
+void SRCAudioConverter::setChannels(int ch)
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("SRCAudioConverter::setChannels this:%p channels:%d\n", this, ch);
+  #endif
+  if(_src_state)
+    src_delete(_src_state);
+  _src_state = 0;
+  
+  _channels = ch;
+  int srcerr;
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("SRCAudioConverter::setChannels Creating samplerate converter type:%d with %d channels\n", _type, ch);
+  #endif  
+  _src_state = src_new(_type, ch, &srcerr);  
+  if(!_src_state)      
+    printf("SRCAudioConverter::setChannels of samplerate converter type:%d with %d channels failed:%s\n", _type, ch, src_strerror(srcerr));
+  return;  
+}
+
+void SRCAudioConverter::reset()
+{
+  if(!_src_state)
+    return;
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("SRCAudioConverter::reset this:%p\n", this);
+  #endif
+  int srcerr = src_reset(_src_state);
+  if(srcerr != 0)      
+    printf("SRCAudioConverter::reset Converter reset failed: %s\n", src_strerror(srcerr));
+  return;  
+}
+
+//off_t SRCAudioConverter::process(SndFileR& f, off_t sfCurFrame, float** buffer, int channel, int n, bool overwrite)
+off_t SRCAudioConverter::process(SndFileR& f, float** buffer, int channel, int n, bool overwrite)
+{
+  //return src_process(_src_state, sd);
+  
+  if(f.isNull())
+    //return;
+    return _sfCurFrame;
+  
+  // Added by Tim. p3.3.17
+  //#ifdef AUDIOCONVERT_DEBUG_PRC
+  //printf("AudioConverter::process %s audConv:%p sfCurFrame:%ld offset:%u channel:%d fchan:%d n:%d\n", 
+  //        f.name().latin1(), this, sfCurFrame, offset, channel, f.channels(), n);
+  //#endif
+  
+//  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);  
+  
+  if((sampleRate == 0) || (fsrate == 0))
+  {  
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("SRCAudioConverter::process Error: sampleRate or file samplerate is zero!\n");
+    #endif
+    return _sfCurFrame;
+  }  
+  
+  SRC_DATA srcdata;
+  int fchan       = f.channels();
+  // Ratio is defined as output sample rate over input samplerate.
+  double srcratio = (double)sampleRate / (double)fsrate;
+  // Extra input compensation.
+  long inComp = 1;
+  
+  long outFrames  = n;  
+  //long outSize   = outFrames * channel;
+  long outSize    = outFrames * fchan;
+  
+  //long inSize = long(outSize * srcratio) + 1                      // From MusE-2 file converter.
+  //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 inSize = inFrames * fchan;
+  //long inSize = inFrames * channel;
+  
+  // Start with buffers at expected sizes. We won't need anything larger than this, but add 4 for good luck.
+  float inbuffer[inSize + 4];
+  float outbuffer[outSize];
+      
+  //size_t sfTotalRead  = 0;
+  size_t rn           = 0;
+  long totalOutFrames = 0;
+  
+  srcdata.data_in       = inbuffer;
+  srcdata.data_out      = outbuffer;
+//  srcdata.data_out      = buffer;
+  
+  // Set some kind of limit on the number of attempts to completely fill the output buffer, 
+  //  in case something is really screwed up - we don't want to get stuck in a loop here.
+  int attempts = 10;
+  for(int attempt = 0; attempt < attempts; ++attempt)
+  {
+    rn = f.readDirect(inbuffer, inFrames);
+    //sfTotalRead += rn;
+    
+    // convert
+    //srcdata.data_in       = inbuffer;
+    //srcdata.data_out      = outbuffer;
+    //srcdata.data_out      = poutbuf;
+    //srcdata.input_frames  = inSize;
+    srcdata.input_frames  = rn;
+    srcdata.output_frames = outFrames;
+    srcdata.end_of_input  = ((long)rn != inFrames);
+    srcdata.src_ratio     = srcratio;
+  
+    //#ifdef AUDIOCONVERT_DEBUG_PRC
+    //printf("AudioConverter::process attempt:%d inFrames:%ld outFrames:%ld rn:%d data in:%p out:%p", 
+    //  attempt, inFrames, outFrames, rn, srcdata.data_in, srcdata.data_out);
+    //#endif
+    
+    int srcerr = src_process(_src_state, &srcdata);
+    if(srcerr != 0)      
+    {
+      printf("\nSRCAudioConverter::process SampleRate converter process failed: %s\n", src_strerror(srcerr));
+      return _sfCurFrame += rn;
+    }
+    
+    totalOutFrames += srcdata.output_frames_gen;
+    
+    //#ifdef AUDIOCONVERT_DEBUG_PRC
+    //printf(" frames used in:%ld out:%ld totalOutFrames:%ld data in:%p out:%p\n", srcdata.input_frames_used, srcdata.output_frames_gen, totalOutFrames, srcdata.data_in, srcdata.data_out);
+    //#endif
+    
+    #ifdef AUDIOCONVERT_DEBUG
+    if(srcdata.output_frames_gen != outFrames)
+      printf("SRCAudioConverter::process %s output_frames_gen:%ld != outFrames:%ld inFrames:%ld srcdata.input_frames_used:%ld rn:%d\n", 
+        f.name().latin1(), srcdata.output_frames_gen, outFrames, inFrames, srcdata.input_frames_used, rn); 
+    #endif
+    
+    // If the number of frames read by the soundfile equals the input frames, go back.
+    // Otherwise we have reached the end of the file, so going back is useless since
+    //  there shouldn't be any further calls. 
+    if((long)rn == inFrames)
+    {
+      // Go back by the amount of unused frames.
+      sf_count_t seekn = inFrames - srcdata.input_frames_used;
+      if(seekn != 0)
+      {
+        #ifdef AUDIOCONVERT_DEBUG_PRC
+        printf("SRCAudioConverter::process Seek-back by:%d\n", seekn);
+        #endif
+        _sfCurFrame = f.seek(-seekn, SEEK_CUR);
+      }
+      else  
+        _sfCurFrame += rn;
+      
+      if(totalOutFrames == n)
+      {
+        // We got our desired number of output frames. Stop attempting.
+        break;
+      }
+      else  
+      {
+        // No point in continuing if on last attempt.
+        if(attempt == (attempts - 1))
+          break;
+          
+        #ifdef AUDIOCONVERT_DEBUG
+        printf("SRCAudioConverter::process %s attempt:%d totalOutFrames:%ld != n:%d try again\n", f.name().latin1(), attempt, totalOutFrames, n);
+        #endif
+        
+        // SRC didn't give us the number of frames we requested. 
+        // This can occasionally be radically different from the requested frames, or zero,
+        //  even when ample excess input frames are supplied.
+        // Move the src output pointer to a new position.
+        srcdata.data_out += srcdata.output_frames_gen * channel;
+        // Set new number of maximum out frames.
+        outFrames -= srcdata.output_frames_gen;
+        // Calculate the new number of file input frames required.
+        inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp;
+        // Keep trying.
+        continue;
+      }  
+    }
+    else
+    {
+      _sfCurFrame += rn;
+      #ifdef AUDIOCONVERT_DEBUG
+      printf("SRCAudioConverter::process %s rn:%zd != inFrames:%ld output_frames_gen:%ld outFrames:%ld srcdata.input_frames_used:%ld\n", 
+        f.name().latin1(), rn, inFrames, srcdata.output_frames_gen, outFrames, srcdata.input_frames_used);
+      #endif
+      
+      // We've reached the end of the file. Convert the number of frames read.
+      //rn = (double)rn * srcratio + 1; 
+      //rn = (long)floor((double)rn * srcratio); 
+      //if(rn > (size_t)outFrames)
+      //  rn = outFrames;  
+      // Stop attempting.
+      break;  
+    }
+  }
+  
+  // If we still didn't get the desired number of output frames.
+  if(totalOutFrames != n)
+  {
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("SRCAudioConverter::process %s totalOutFrames:%ld != n:%d\n", f.name().latin1(), totalOutFrames, n);
+    #endif
+          
+    // Let's zero the rest of it.
+    long b = totalOutFrames * channel;
+    long e = n * channel;
+    for(long i = b; i < e; ++i)
+      outbuffer[i] = 0.0f;
+      //buffer[i] = 0.0f;
+  }
+  
+  //float*  poutbuf = &outbuffer[0];
+  float*  poutbuf = outbuffer;
+  if(fchan == channel) 
+  {
+    if(overwrite)
+      //for (size_t i = 0; i < rn; ++i) 
+      for (int i = 0; i < n; ++i) 
+      {
+        for(int ch = 0; ch < channel; ++ch)
+          *(buffer[ch] + i) = *poutbuf++;
+      }
+    else
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        for(int ch = 0; ch < channel; ++ch)
+          *(buffer[ch] + i) += *poutbuf++;
+      }
+  }
+  else if((fchan == 2) && (channel == 1)) 
+  {
+    // stereo to mono
+    if(overwrite)
+      //for(size_t i = 0; i < rn; ++i)
+      for(int i = 0; i < n; ++i)
+        *(buffer[0] + i) = poutbuf[i + i] + poutbuf[i + i + 1];
+    else  
+      //for(size_t i = 0; i < rn; ++i)
+      for(int i = 0; i < n; ++i)
+        *(buffer[0] + i) += poutbuf[i + i] + poutbuf[i + i + 1];
+  }
+  else if((fchan == 1) && (channel == 2)) 
+  {
+    // mono to stereo
+    if(overwrite)
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        float data = *poutbuf++;
+        *(buffer[0]+i) = data;
+        *(buffer[1]+i) = data;
+      }
+    else  
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        float data = *poutbuf++;
+        *(buffer[0]+i) += data;
+        *(buffer[1]+i) += data;
+      }
+  }
+  else 
+  {
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("SRCAudioConverter::process Channel mismatch: source chans:%d -> dst chans:%d\n", fchan, channel);
+    #endif
+  }
+  
+  return _sfCurFrame;
+}
+
+#ifdef RUBBERBAND_SUPPORT
+
+//---------------------------------------------------------
+//   RubberBandAudioConverter
+//---------------------------------------------------------
+
+RubberBandAudioConverter::RubberBandAudioConverter(int channels, int options) : AudioConverter()
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("RubberBandAudioConverter::RubberBandAudioConverter this:%p channels:%d options:%x\n", this, channels, options);
+  #endif
+
+  _options = options;
+  _rbs = 0;
+  _channels = channels;
+  
+  _rbs = new RubberBandStretcher(sampleRate, _channels, _options);  // , initialTimeRatio = 1.0, initialPitchScale = 1.0
+}
+
+RubberBandAudioConverter::~RubberBandAudioConverter()
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("RubberBandAudioConverter::~RubberBandAudioConverter this:%p\n", this);
+  #endif
+  if(_rbs)
+    delete _rbs;
+}
+
+void RubberBandAudioConverter::setChannels(int ch)
+{
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("RubberBandAudioConverter::setChannels this:%p channels:%d\n", this, ch);
+  #endif
+  if(_rbs)
+    delete _rbs;
+  _rbs = 0;
+  
+  _channels = ch;
+  _rbs = new RubberBandStretcher(sampleRate, _channels, _options);  // , initialTimeRatio = 1.0, initialPitchScale = 1.0
+}
+
+void RubberBandAudioConverter::reset()
+{
+  if(!_rbs)
+    return;
+  #ifdef AUDIOCONVERT_DEBUG
+  printf("RubberBandAudioConverter::reset this:%p\n", this);
+  #endif
+  _rbs->reset();
+  return;  
+}
+
+/////////////////////////////////
+// TODO: Not finished yet..
+////////////////////////////////
+//off_t RubberBandAudioConverter::process(SndFileR& f, off_t sfCurFrame, float** buffer, int channel, int n, bool overwrite)
+off_t RubberBandAudioConverter::process(SndFileR& f, float** buffer, int channel, int n, bool overwrite)
+{
+  //return src_process(_src_state, sd);
+  
+  if(f.isNull())
+    //return;
+    return _sfCurFrame;
+  
+  // Added by Tim. p3.3.17
+  //#ifdef AUDIOCONVERT_DEBUG_PRC
+  //printf("AudioConverter::process %s audConv:%p sfCurFrame:%ld offset:%u channel:%d fchan:%d n:%d\n", 
+  //        f.name().latin1(), this, sfCurFrame, offset, channel, f.channels(), n);
+  //#endif
+  
+//  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);  
+  
+  if((sampleRate == 0) || (fsrate == 0))
+  {  
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("RubberBandAudioConverter::process Error: sampleRate or file samplerate is zero!\n");
+    #endif
+    return _sfCurFrame;
+  }  
+  
+//  SRC_DATA srcdata;
+  int fchan       = f.channels();
+  // Ratio is defined as output sample rate over input samplerate.
+  double srcratio = (double)sampleRate / (double)fsrate;
+  // Extra input compensation.
+  long inComp = 1;
+  
+  long outFrames  = n;  
+  //long outSize   = outFrames * channel;
+  long outSize    = outFrames * fchan;
+  
+  //long inSize = long(outSize * srcratio) + 1                      // From MusE-2 file converter.
+  //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 inSize = inFrames * fchan;
+  //long inSize = inFrames * channel;
+  
+  // Start with buffers at expected sizes. We won't need anything larger than this, but add 4 for good luck.
+  float inbuffer[inSize]; // +4
+//  float outbuffer[outSize];
+      
+  //float* rbinbuffer[fchan];
+  //float rbindata[inSize];
+  //for (int i = 0; i < fchan; ++i)
+  //      rbinbuffer[i] = rbindata + i * inFrames;
+      
+  float* rboutbuffer[fchan];
+  float rboutdata[outSize];
+  for (int i = 0; i < fchan; ++i)
+        rboutbuffer[i] = rboutdata + i * outFrames;
+      
+  //size_t sfTotalRead  = 0;
+  size_t rn           = 0;
+  long totalOutFrames = 0;
+  
+//  srcdata.data_in       = inbuffer;
+  //srcdata.data_out      = outbuffer;
+//  srcdata.data_out      = buffer;
+  float** data_out       = rboutbuffer;
+  
+  // For just sample rate conversion, apply same ratio to both time and pitch.
+  _rbs->setTimeRatio(srcratio);
+  _rbs->setPitchScale(srcratio);
+  
+  // Set some kind of limit on the number of attempts to completely fill the output buffer, 
+  //  in case something is really screwed up - we don't want to get stuck in a loop here.
+  int attempts = 10;
+  for(int attempt = 0; attempt < attempts; ++attempt)
+  {
+    size_t sreq = _rbs->getSamplesRequired();
+    
+    size_t rbinSize = sreq * fchan;
+    float* rbinbuffer[fchan];
+    float rbindata[rbinSize];
+    for(int i = 0; i < fchan; ++i)
+      rbinbuffer[i] = rbindata + i * sreq;
+    
+//    rn = f.readDirect(inbuffer, inFrames);
+    rn = f.readDirect(inbuffer, sreq);
+    //sfTotalRead += rn;
+    
+    // Must de-interleave soundfile data to feed to rubberband.
+    for(size_t i = 0; i < rn; ++i) 
+    {
+      for(int ch = 0; ch < fchan; ++ch)
+        *(rbinbuffer[ch] + i) = *inbuffer++;
+    }
+           
+    _rbs->process(rbinbuffer, rn, (long)rn != inFrames);
+    
+    // "This function returns -1 if all data has been fully processed and all output read, and the stretch process is now finished." 
+    int savail = _rbs->available(); 
+    
+    
+    // convert
+    //srcdata.data_in       = inbuffer;
+    //srcdata.data_out      = outbuffer;
+    //srcdata.data_out      = poutbuf;
+    //srcdata.input_frames  = inSize;
+    srcdata.input_frames  = rn;
+    srcdata.output_frames = outFrames;
+    srcdata.end_of_input  = ((long)rn != inFrames);
+    srcdata.src_ratio     = srcratio;
+  
+    //#ifdef AUDIOCONVERT_DEBUG_PRC
+    //printf("AudioConverter::process attempt:%d inFrames:%ld outFrames:%ld rn:%d data in:%p out:%p", 
+    //  attempt, inFrames, outFrames, rn, srcdata.data_in, srcdata.data_out);
+    //#endif
+    
+    int srcerr = src_process(_src_state, &srcdata);
+    if(srcerr != 0)      
+    {
+      printf("\RubberBandAudioConverter::process SampleRate converter process failed: %s\n", src_strerror(srcerr));
+      return _sfCurFrame += rn;
+    }
+    
+    totalOutFrames += srcdata.output_frames_gen;
+    
+    //#ifdef AUDIOCONVERT_DEBUG_PRC
+    //printf(" frames used in:%ld out:%ld totalOutFrames:%ld data in:%p out:%p\n", srcdata.input_frames_used, srcdata.output_frames_gen, totalOutFrames, srcdata.data_in, srcdata.data_out);
+    //#endif
+    
+    #ifdef AUDIOCONVERT_DEBUG
+    if(srcdata.output_frames_gen != outFrames)
+      printf("RubberBandAudioConverter::process %s output_frames_gen:%ld != outFrames:%ld inFrames:%ld srcdata.input_frames_used:%ld rn:%d\n", 
+        f.name().latin1(), srcdata.output_frames_gen, outFrames, inFrames, srcdata.input_frames_used, rn); 
+    #endif
+    
+    // If the number of frames read by the soundfile equals the input frames, go back.
+    // Otherwise we have reached the end of the file, so going back is useless since
+    //  there shouldn't be any further calls. 
+    if((long)rn == inFrames)
+    {
+      // Go back by the amount of unused frames.
+      sf_count_t seekn = inFrames - srcdata.input_frames_used;
+      if(seekn != 0)
+      {
+        #ifdef AUDIOCONVERT_DEBUG_PRC
+        printf("RubberBandAudioConverter::process Seek-back by:%d\n", seekn);
+        #endif
+        _sfCurFrame = f.seek(-seekn, SEEK_CUR);
+      }
+      else  
+        _sfCurFrame += rn;
+      
+      if(totalOutFrames == n)
+      {
+        // We got our desired number of output frames. Stop attempting.
+        break;
+      }
+      else  
+      {
+        // No point in continuing if on last attempt.
+        if(attempt == (attempts - 1))
+          break;
+          
+        #ifdef AUDIOCONVERT_DEBUG
+        printf("RubberBandAudioConverter::process %s attempt:%d totalOutFrames:%ld != n:%d try again\n", f.name().latin1(), attempt, totalOutFrames, n);
+        #endif
+        
+        // We didn't get the number of frames we requested. 
+        // This can occasionally be radically different from the requested frames, or zero,
+        //  even when ample excess input frames are supplied.
+        // Move the src output pointer to a new position.
+        srcdata.data_out += srcdata.output_frames_gen * channel;
+        // Set new number of maximum out frames.
+        outFrames -= srcdata.output_frames_gen;
+        // Calculate the new number of file input frames required.
+        inFrames = (long)ceil(((double)outFrames / srcratio)) + inComp;
+        // Keep trying.
+        continue;
+      }  
+    }
+    else
+    {
+      _sfCurFrame += rn;
+      #ifdef AUDIOCONVERT_DEBUG
+      printf("RubberBandAudioConverter::process %s rn:%zd != inFrames:%ld output_frames_gen:%ld outFrames:%ld srcdata.input_frames_used:%ld\n", 
+        f.name().latin1(), rn, inFrames, srcdata.output_frames_gen, outFrames, srcdata.input_frames_used);
+      #endif
+      
+      // We've reached the end of the file. Convert the number of frames read.
+      //rn = (double)rn * srcratio + 1; 
+      //rn = (long)floor((double)rn * srcratio); 
+      //if(rn > (size_t)outFrames)
+      //  rn = outFrames;  
+      // Stop attempting.
+      break;  
+    }
+  }
+  
+  // If we still didn't get the desired number of output frames.
+  if(totalOutFrames != n)
+  {
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("RubberBandAudioConverter::process %s totalOutFrames:%ld != n:%d\n", f.name().latin1(), totalOutFrames, n);
+    #endif
+          
+    // Let's zero the rest of it.
+    long b = totalOutFrames * channel;
+    long e = n * channel;
+    for(long i = b; i < e; ++i)
+      //outbuffer[i] = 0.0f;
+      buffer[i] = 0.0f;
+  }
+  
+  //float*  poutbuf = &outbuffer[0];
+  float*  poutbuf = outbuffer;
+  if(fchan == channel) 
+  {
+    if(overwrite)
+      //for (size_t i = 0; i < rn; ++i) 
+      for (int i = 0; i < n; ++i) 
+      {
+        for(int ch = 0; ch < channel; ++ch)
+          *(buffer[ch] + i) = *poutbuf++;
+      }
+    else
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        for(int ch = 0; ch < channel; ++ch)
+          *(buffer[ch] + i) += *poutbuf++;
+      }
+  }
+  else if((fchan == 2) && (channel == 1)) 
+  {
+    // stereo to mono
+    if(overwrite)
+      //for(size_t i = 0; i < rn; ++i)
+      for(int i = 0; i < n; ++i)
+        *(buffer[0] + i) = poutbuf[i + i] + poutbuf[i + i + 1];
+    else  
+      //for(size_t i = 0; i < rn; ++i)
+      for(int i = 0; i < n; ++i)
+        *(buffer[0] + i) += poutbuf[i + i] + poutbuf[i + i + 1];
+  }
+  else if((fchan == 1) && (channel == 2)) 
+  {
+    // mono to stereo
+    if(overwrite)
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        float data = *poutbuf++;
+        *(buffer[0]+i) = data;
+        *(buffer[1]+i) = data;
+      }
+    else  
+      //for(size_t i = 0; i < rn; ++i) 
+      for(int i = 0; i < n; ++i) 
+      {
+        float data = *poutbuf++;
+        *(buffer[0]+i) += data;
+        *(buffer[1]+i) += data;
+      }
+  }
+  else 
+  {
+    #ifdef AUDIOCONVERT_DEBUG
+    printf("RubberBandAudioConverter::process Channel mismatch: source chans:%d -> dst chans:%d\n", fchan, channel);
+    #endif
+  }
+  
+  return _sfCurFrame;
+}
+
+#endif // RUBBERBAND_SUPPORT