/home/marsyas/marsyas/src/marsyas/AudioSource.cpp

/*
** Copyright (C) 1998-2005 George Tzanetakis <gtzan@cs.cmu.edu>
**  
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
** 
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
** GNU General Public License for more details.
** 
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software 
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/

#include "common.h" 
#include "AudioSource.h"

using namespace std;
using namespace Marsyas;

AudioSource::AudioSource(string name):MarSystem("AudioSource", name)
{

#ifdef MARSYAS_AUDIOIO
    audio_ = NULL;
#endif


  ri_ = 0;
  pringBufferSize_ = 0;

  isInitialized_ = false;
  stopped_ = true;

  rtSrate_ = 0;
  bufferSize_ = 0;
  rtChannels_ = 0;
  nChannels_ = 0;
  pnChannels_ = 0;
  
  addControls();
}

AudioSource::~AudioSource()
{
#ifdef MARSYAS_AUDIOIO
    delete audio_;
#endif 

}


MarSystem* 
AudioSource::clone() const
{
  return new AudioSource(*this);
}

void 
AudioSource::addControls()
{
  addctrl("mrs_natural/nChannels", 1);

  
#ifdef MARSYAS_MACOSX
  addctrl("mrs_natural/bufferSize", 1024);
#else
  addctrl("mrs_natural/bufferSize", 256);
#endif

  
  addctrl("mrs_bool/initAudio", false);
  setctrlState("mrs_bool/initAudio", true);
  
  addctrl("mrs_bool/hasData", true);
  addctrl("mrs_real/gain", 1.0); 
}




int 
AudioSource::recordCallback(void *outputBuffer, void *inputBuffer, 
                            unsigned int nBufferFrames, 
                            double streamTime, unsigned int status, 
                            void *userData)
{
    
    unsigned int drain_count = 0;
    mrs_real* data = (mrs_real*)inputBuffer;
    InputData *iData = (InputData *)userData;
    AudioSource* mythis = (AudioSource *)iData->myself;
    realvec& ringBuffer = *(iData->ringBuffer);
    unsigned int t;
    unsigned int spaceAvailable;
    
    for (t=0; t < nBufferFrames; t++)
    {
        if (iData->samplesInBuffer <= iData->high_watermark)
        {
            ringBuffer(0,iData->wp) = data[2*t];            
            ringBuffer(1,iData->wp) = data[2*t+1];
            
            iData->wp = ++ (iData->wp) % iData->ringBufferSize;
            if (iData->wp >= iData->rp) 
                iData->samplesInBuffer = iData->wp - iData->rp;
            else 
                iData->samplesInBuffer = iData->ringBufferSize - (iData->rp - iData->wp);
        }
        else
        {
            while (iData->samplesInBuffer > iData->high_watermark)
            {
                SLEEP(1);
                drain_count++;
                if (drain_count == 1000)
                    return 1;
            }
        }
    }
    return 0;
}



void 
AudioSource::myUpdate(MarControlPtr sender)
{
    (void) sender;
  MRSDIAG("AudioSource::myUpdate");
  
  //set output controls
  setctrl("mrs_natural/onSamples", getctrl("mrs_natural/inSamples"));
  setctrl("mrs_real/osrate", getctrl("mrs_real/israte"));
  setctrl("mrs_natural/onObservations", getctrl("mrs_natural/nChannels"));
  
  
  inSamples_ = getctrl("mrs_natural/inSamples")->to<mrs_natural>();
  inObservations_ = ctrl_inObservations_->to<mrs_natural>();
  gain_ = getctrl("mrs_real/gain")->to<mrs_real>();
  


  //resize ringBuffer if necessary
  if (inSamples_ < bufferSize_) 
      ringBufferSize_ =  8 * bufferSize_;
  else 
  {
      ringBufferSize_ =  8 * inSamples_;
  }
  

  idata.ringBufferSize = ringBufferSize_;
  idata.high_watermark = ringBufferSize_/4;
  idata.low_watermark = ringBufferSize_/8;
  

  nChannels_ = getctrl("mrs_natural/nChannels")->to<mrs_natural>(); 
  if ((ringBufferSize_ > pringBufferSize_)||(nChannels_ != pnChannels_))
    {
        ringBuffer_.stretch(nChannels_, ringBufferSize_);
    }
  pringBufferSize_ = ringBufferSize_;
  pnChannels_ = nChannels_;
  


  
  
  if (getctrl("mrs_bool/initAudio")->to<mrs_bool>()) 
      initRtAudio();
}


void 
AudioSource::initRtAudio()
{

    bufferSize_ = (int)getctrl("mrs_natural/bufferSize")->to<mrs_natural>();
    nChannels_ = getctrl("mrs_natural/nChannels")->to<mrs_natural>();
    rtSrate_ = (int)getctrl("mrs_real/israte")->to<mrs_real>();
    rtChannels_ = (int)getctrl("mrs_natural/nChannels")->to<mrs_natural>();

    
    //marsyas represents audio data as float numbers
#ifdef MARSYAS_AUDIOIO
    if (audio_ == NULL)
        audio_ = new RtAudio();
    

    unsigned int bufferFrames = bufferSize_;
    
    
    RtAudio::StreamParameters iParams;
    iParams.deviceId = audio_->getDefaultInputDevice();
    iParams.nChannels = rtChannels_;
    iParams.firstChannel = 0; 
    
    idata.ringBuffer = &ringBuffer_;
    idata.wp = 0;
    idata.rp = 0;
    idata.ringBufferSize = ringBufferSize_;
    idata.myself = this;
    

    


    RtAudioFormat rtFormat = (sizeof(mrs_real) == 8) ? RTAUDIO_FLOAT64 : RTAUDIO_FLOAT32;
    
    
    try 
    {
        audio_->openStream(NULL, &iParams, rtFormat, rtSrate_, 
                           &bufferFrames, &recordCallback, (void *)&idata, NULL);
    }
    catch (RtError& e) 
    {
        e.printMessage();
        exit(0);
    }
    

  //update bufferSize control which may have been changed
  //by RtAudio (see RtAudio documentation)
    setctrl("mrs_natural/bufferSize", (mrs_natural)bufferFrames);

#endif 

  isInitialized_ = true;
  setctrl("mrs_bool/initAudio", false);
}

void 
AudioSource::start()
{
#ifdef MARSYAS_AUDIOIO
    if ( stopped_) {
        audio_->startStream();
        stopped_ = false;
    }
#endif 
}

void 
AudioSource::stop()
{
#ifdef MARSYAS_AUDIOIO
  if ( !stopped_ ) {
    audio_->stopStream();
    stopped_ = true;
  }
#endif 
}

void
AudioSource::localActivate(bool state)
{
  if(state)
    start();
  else
    stop();
}


unsigned int 
AudioSource::getSpaceAvailable() 
{
    unsigned int free = (idata.rp - idata.wp -1 + idata.ringBufferSize) % idata.ringBufferSize;
    unsigned int underMark = idata.high_watermark - getSamplesAvailable();
    
    return(min(underMark, free));
}


unsigned int 
AudioSource::getSamplesAvailable() 
{
    unsigned int samplesAvailable = (idata.wp - idata.rp +idata.ringBufferSize) % idata.ringBufferSize;
    return samplesAvailable;
}


void 
AudioSource::myProcess(realvec& in, realvec& out)
{
    
    
    (void) in;
    
    //check if RtAudio is initialized
    if (!isInitialized_)
        return;
    
    //check MUTE
    if(ctrl_mute_->isTrue()) return;

    //assure that RtAudio thread is running
    //(this may be needed by if an explicit call to start()
    //is not done before ticking or calling process() )
    if ( stopped_ )
        start();
    
    for (t=0; t < onSamples_; t++)
    {
        if (getSamplesAvailable()) 
        {
            for (o=0; o < onObservations_; o++)           
                out(o,t) = ringBuffer_(o,idata.rp);
            idata.rp = ++(idata.rp) % idata.ringBufferSize; 
            if (idata.rp >= idata.rp) 
                idata.samplesInBuffer = idata.wp - idata.rp;
            else 
                idata.samplesInBuffer = idata.ringBufferSize - (idata.rp - idata.wp);
          
        }
    }
    
    while (idata.samplesInBuffer < idata.low_watermark)
    {
        SLEEP(1);
    }
    
    
    
    
}

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