audacia/src/AudioIO.cpp

/**********************************************************************

  Audacity: A Digital Audio Editor

  AudioIO.cpp

  Copyright 2000-2004:
  Dominic Mazzoni
  Joshua Haberman
  Markus Meyer
  Matt Brubeck

  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.

********************************************************************//**

\class AudioIO
\brief AudioIO uses the PortAudio library to play and record sound.

  Great care and attention to detail are necessary for understanding and
  modifying this system.  The code in this file is run from three
  different thread contexts: the UI thread, the disk thread (which
  this file creates and maintains) and the PortAudio callback thread.
  To highlight this deliniation, the file is divided into three parts
  based on what thread context each function is intended to run in.

  \todo run through all functions called from audio and portaudio threads
  to verify they are thread-safe.

*//****************************************************************//**

\class AudioThread
\brief Defined different on Mac and other platforms (on Mac it does not
use wxWidgets wxThread), this class sits in a thread loop reading and 
writing audio.

*//*******************************************************************/

#include "Audacity.h"
#include "float_cast.h"
#include "Experimental.h"

#include <math.h>
#include <stdlib.h>
#include <algorithm>

#ifdef __WXMSW__
#include <malloc.h>
#endif

#ifdef HAVE_ALLOCA_H
#include <alloca.h>
#endif

#if USE_PORTMIXER
#include "portmixer.h"
#endif

#include <wx/log.h>
#include <wx/textctrl.h>
#include <wx/msgdlg.h>
#include <wx/timer.h>
#include <wx/intl.h>
#include <wx/debug.h>
#include <wx/sstream.h>
#include <wx/txtstrm.h>

#include "AudacityApp.h"
#include "AudioIO.h"
#include "WaveTrack.h"

#ifdef EXPERIMENTAL_MIDI_OUT
#define MIDI_BUFFER_AHEAD 1.0 /* secondds */
#define ROUND(x) (int) ((x)+0.5)
//#include <string.h>
#include "portmidi.h"
#include "NoteTrack.h"
#endif

#include "Mix.h"
#include "Resample.h"
#include "RingBuffer.h"
#include "Prefs.h"
#include "Project.h"
#include "toolbars/ControlToolBar.h"

#include "widgets/Meter.h"
#include "../Experimental.h"

#define NO_STABLE_INDICATOR -1000000000
#define LOWER_BOUND 0.0
#define UPPER_BOUND 1.0

using std::max;
using std::min;

AudioIO *gAudioIO;

// static
int AudioIO::mNextStreamToken = 0;
int AudioIO::mCachedPlaybackIndex = -1;
wxArrayLong AudioIO::mCachedPlaybackRates;
int AudioIO::mCachedCaptureIndex = -1;
wxArrayLong AudioIO::mCachedCaptureRates;
wxArrayLong AudioIO::mCachedSampleRates;
double AudioIO::mCachedBestRateIn = 0.0;
double AudioIO::mCachedBestRateOut;

const int AudioIO::StandardRates[] = {
   8000,
   11025,
   16000,
   22050,
   32000,
   44100,
   48000,
   96000
};
const int AudioIO::NumStandardRates = sizeof(AudioIO::StandardRates) /
                                      sizeof(AudioIO::StandardRates[0]);
const int AudioIO::RatesToTry[] = {
   8000,
   9600,
   11025,
   12000,
   15000,
   16000,
   22050,
   24000,
   32000,
   44100,
   48000,
   88200,
   96000,
   192000
};
const int AudioIO::NumRatesToTry = sizeof(AudioIO::RatesToTry) /
                                      sizeof(AudioIO::RatesToTry[0]);

int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
                          unsigned long framesPerBuffer,
                          const PaStreamCallbackTimeInfo *timeInfo,
                          PaStreamCallbackFlags statusFlags, void *userData );

#ifdef EXPERIMENTAL_MIDI_OUT
int compareTime( const void* a, const void* b );
#endif

//////////////////////////////////////////////////////////////////////
//
//     class AudioThread - declaration and glue code
//
//////////////////////////////////////////////////////////////////////

#ifdef __WXMAC__

// On Mac OS X, it's better not to use the wxThread class.
// We use our own implementation based on pthreads instead.

#include <pthread.h>
#include <time.h>

class AudioThread {
 public:
   typedef int ExitCode;
   AudioThread() { mDestroy = false; mThread = NULL; }
   ExitCode Entry();
   void Create() {}
   void Delete() {
      mDestroy = true;
      pthread_join(mThread, NULL);
   }
   bool TestDestroy() { return mDestroy; }
   void Sleep(int ms) {
      struct timespec spec;
      spec.tv_sec = 0;
      spec.tv_nsec = ms * 1000 * 1000;
      nanosleep(&spec, NULL);
   }
   static void *callback(void *p) {
      AudioThread *th = (AudioThread *)p;
      return (void *)th->Entry();
   }
   void Run() {
      pthread_create(&mThread, NULL, callback, this);
   }
 private:
   bool mDestroy;
   pthread_t mThread;

};

#else

// The normal wxThread-derived AudioThread class for all other
// platforms:
class AudioThread : public wxThread {
 public:
   AudioThread():wxThread(wxTHREAD_JOINABLE) {}
   virtual ExitCode Entry();
};

#endif

//////////////////////////////////////////////////////////////////////
//
//     UI Thread Context
//
//////////////////////////////////////////////////////////////////////

void InitAudioIO()
{
   gAudioIO = new AudioIO();
   gAudioIO->mThread->Run();

   // Make sure device prefs are initialized
   if (gPrefs->Read(wxT("AudioIO/RecordingDevice"), wxT("")) == wxT("")) {
      int i = AudioIO::getRecordDevIndex();
      const PaDeviceInfo *info = Pa_GetDeviceInfo(i);
      if (info) {
         gPrefs->Write(wxT("/AudioIO/RecordingDevice"), DeviceName(info));
         gPrefs->Write(wxT("/AudioIO/Host"),
               wxString(Pa_GetHostApiInfo(info->hostApi)->name, wxConvLocal));
      }
   }

   if (gPrefs->Read(wxT("AudioIO/PlaybackDevice"), wxT("")) == wxT("")) {
      int i = AudioIO::getPlayDevIndex();
      const PaDeviceInfo *info = Pa_GetDeviceInfo(i);
      if (info) {
         gPrefs->Write(wxT("/AudioIO/PlaybackDevice"), DeviceName(info));
         gPrefs->Write(wxT("/AudioIO/Host"),
               wxString(Pa_GetHostApiInfo(info->hostApi)->name, wxConvLocal));
      }
   }
}

void DeinitAudioIO()
{
   delete gAudioIO;
}

wxString DeviceName(const PaDeviceInfo* info)
{
   wxString hostapiName(Pa_GetHostApiInfo(info->hostApi)->name, wxConvLocal);
   wxString infoName(info->name, wxConvLocal);

   return wxString::Format(wxT("%s: %s"),
                           hostapiName.c_str(),
                           infoName.c_str());
}

bool AudioIO::ValidateDeviceNames(wxString play, wxString rec)
{
   const PaDeviceInfo *pInfo = Pa_GetDeviceInfo(AudioIO::getPlayDevIndex(play));
   const PaDeviceInfo *rInfo = Pa_GetDeviceInfo(AudioIO::getRecordDevIndex(rec));

   if (!pInfo || !rInfo || pInfo->hostApi != rInfo->hostApi) {
      return false;
   }

   return true;
}

AudioIO::AudioIO()
{
   mAudioThreadShouldCallFillBuffersOnce = false;
   mAudioThreadFillBuffersLoopRunning = false;
   mAudioThreadFillBuffersLoopActive = false;
   mPortStreamV19 = NULL;

#ifdef EXPERIMENTAL_MIDI_OUT
   mMidiStream = NULL;
   mMidiStreamActive = false;
   mSendMidiState = false;

   mIterator = new Alg_iterator(mSeq, true);
   mIterator->begin(true);
   mNextEvent = mIterator->next();

   mAudioCallbackSampleNumber = 0;
#endif

#ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   mAILAActive = false;
#endif
   mSilentBuf = NULL;
   mLastSilentBufSize = 0;

   mStreamToken = 0;
   mStopStreamCount = 0;
   mTempFloats = new float[65536]; // TODO: out channels * PortAudio buffer size

   mLastPaError = paNoError;

   mLastRecordingOffset = 0.0;
   mNumCaptureChannels = 0;
   mPaused = false;
   mPlayLooped = false;

   mListener = NULL;
   mUpdateMeters = false;
   mUpdatingMeters = false;

   PaError err = Pa_Initialize();

   if (err != paNoError) {
      wxString errStr = _("Could not find any audio devices.\n");
      errStr += _("You will not be able to play or record audio.\n\n");
      wxString paErrStr = LAT1CTOWX(Pa_GetErrorText(err));
      if (paErrStr)
         errStr += _("Error: ")+paErrStr;
      // XXX: we are in libaudacity, popping up dialogs not allowed!  A
      // long-term solution will probably involve exceptions
      wxMessageBox(errStr, _("Error Initializing Audio"), wxICON_ERROR|wxOK);

      // Since PortAudio is not initialized, all calls to PortAudio
      // functions will fail.  This will give reasonable behavior, since
      // the user will be able to do things not relating to audio i/o,
      // but any attempt to play or record will simply fail.
   }

#ifdef EXPERIMENTAL_MIDI_OUT
   PmError pmErr = Pm_Initialize();

   if (pmErr != pmNoError) {
      wxString errStr = 
              _("There was an error initializing the midi i/o layer.\n");
      errStr += _("You will not be able to play midi.\n\n");
      wxString pmErrStr = LAT1CTOWX(Pm_GetErrorText(pmErr));
      if (pmErrStr)
         errStr += _("Error: ") + pmErrStr;
      // XXX: we are in libaudacity, popping up dialogs not allowed!  A
      // long-term solution will probably involve exceptions
	  wxMessageBox(errStr, _("Error Initializing Midi"), wxICON_ERROR|wxOK);

      // Same logic for PortMidi as described above for PortAudio
   }
#endif

   // Start thread
   mThread = new AudioThread();
   mThread->Create();

#if defined(USE_PORTMIXER)
   mPortMixer = NULL;
   mPreviousHWPlaythrough = -1.0;
   HandleDeviceChange();
#else
   mEmulateMixerOutputVol = true;
   mMixerOutputVol = 1.0;
   mInputMixerWorks = false;
#endif
}

AudioIO::~AudioIO()
{
#if defined(USE_PORTMIXER)
   if (mPortMixer) {
      #if __WXMAC__
      if (Px_SupportsPlaythrough(mPortMixer) && mPreviousHWPlaythrough >= 0.0)
         Px_SetPlaythrough(mPortMixer, mPreviousHWPlaythrough);
         mPreviousHWPlaythrough = -1.0;
      #endif
      Px_CloseMixer(mPortMixer);
      mPortMixer = NULL;
   }
#endif
   Pa_Terminate();

#ifdef EXPERIMENTAL_MIDI_OUT
   Pm_Terminate();
#endif

   /* Delete is a "graceful" way to stop the thread.
      (Kill is the not-graceful way.) */
   wxYield();
   mThread->Delete();
   
   if(mSilentBuf) 
      DeleteSamples(mSilentBuf);

   delete [] mTempFloats;
   delete mThread;
}

void AudioIO::SetMixer(int recordDevice, float recordVolume,
                       float playbackVolume)
{
   mMixerOutputVol = playbackVolume;

#if defined(USE_PORTMIXER)
   PxMixer *mixer = mPortMixer;

   if( mixer )
   {
      int oldRecordDevice = Px_GetCurrentInputSource(mixer);
      float oldRecordVolume = Px_GetInputVolume(mixer);
      float oldPlaybackVolume = Px_GetPCMOutputVolume(mixer);

      if( recordDevice != oldRecordDevice )
         Px_SetCurrentInputSource(mixer, recordDevice);
      if( oldRecordVolume != recordVolume )
         Px_SetInputVolume(mixer, recordVolume);
      if( oldPlaybackVolume != playbackVolume )
         Px_SetPCMOutputVolume(mixer, playbackVolume);

      return;
   }
#endif
}

void AudioIO::GetMixer(int *recordDevice, float *recordVolume,
                       float *playbackVolume)
{
#if defined(USE_PORTMIXER)

   PxMixer *mixer = mPortMixer;

   if( mixer )
   {
      *recordDevice = Px_GetCurrentInputSource(mixer);

      if (mInputMixerWorks)
         *recordVolume = Px_GetInputVolume(mixer);
      else
         *recordVolume = 1.0f;

      if (mEmulateMixerOutputVol)
         *playbackVolume = mMixerOutputVol;
      else
         *playbackVolume = Px_GetPCMOutputVolume(mixer);

      return;
   }

#endif

   *recordDevice = 0;
   *recordVolume = 1.0f;
   *playbackVolume = mMixerOutputVol;
}

bool AudioIO::InputMixerWorks()
{
   return mInputMixerWorks;
}

wxArrayString AudioIO::GetInputSourceNames()
{
#if defined(USE_PORTMIXER)

   wxArrayString deviceNames;

   if( mPortMixer )
   {
      int numSources = Px_GetNumInputSources(mPortMixer);
      for( int source = 0; source < numSources; source++ )
         deviceNames.Add(wxString(Px_GetInputSourceName(mPortMixer, source), wxConvLocal));
   }
   else
   {
      wxLogDebug(wxT("AudioIO::GetInputSourceNames(): PortMixer not initialised!"));
   }

   return deviceNames;

#else

   wxArrayString blank;

   return blank;

#endif
}

void AudioIO::HandleDeviceChange()
{
   // This should not happen, but it would screw things up if it did.
   if (IsStreamActive())
      return;

   // get the selected record and playback devices
   int playDeviceNum = getPlayDevIndex();
   int recDeviceNum = getRecordDevIndex();

   // cache playback/capture rates
   mCachedPlaybackRates = GetSupportedPlaybackRates(playDeviceNum);
   mCachedCaptureRates = GetSupportedCaptureRates(recDeviceNum);
   mCachedSampleRates = GetSupportedSampleRates(playDeviceNum, recDeviceNum);
   mCachedPlaybackIndex = playDeviceNum;
   mCachedCaptureIndex = recDeviceNum;
   mCachedBestRateIn = 0.0;

#if defined(USE_PORTMIXER)

   // if we have a PortMixer object, close it down
   if (mPortMixer) {
      #if __WXMAC__
      // on the Mac we must make sure that we restore the hardware playthrough
      // state of the sound device to what it was before, because there isn't
      // a UI for this (!)
      if (Px_SupportsPlaythrough(mPortMixer) && mPreviousHWPlaythrough >= 0.0)
         Px_SetPlaythrough(mPortMixer, mPreviousHWPlaythrough);
         mPreviousHWPlaythrough = -1.0;
      #endif
      Px_CloseMixer(mPortMixer);
      mPortMixer = NULL;
   }

   // that might have given us no rates whatsoever, so we have to guess an
   // answer to do the next bit
   int numrates = mCachedSampleRates.GetCount();
   int highestSampleRate;
   if (numrates > 0)
   {
      highestSampleRate = mCachedSampleRates[numrates - 1];
   }
   else
   {  // we don't actually have any rates that work for Rec and Play. Guess one
      // to use for messing with the mixer, which doesn't actually do either
      highestSampleRate = 44100;
      // mCachedSampleRates is still empty, but it's not used again, so
      // can ignore
   }
   mInputMixerWorks = false;
   mEmulateMixerOutputVol = true;
   mMixerOutputVol = 1.0;

   int error;
   // This tries to open the device with the samplerate worked out above, which
   // will be the highest available for play and record on the device, or
   // 44.1kHz if the info cannot be fetched.

   PaStream *stream;

   PaStreamParameters playbackParameters;

   playbackParameters.device = playDeviceNum;
   playbackParameters.sampleFormat = paFloat32;
   playbackParameters.hostApiSpecificStreamInfo = NULL;
   playbackParameters.channelCount = 1;
   if (Pa_GetDeviceInfo(playDeviceNum))
      playbackParameters.suggestedLatency =
         Pa_GetDeviceInfo(playDeviceNum)->defaultLowOutputLatency;
   else
      playbackParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0; 

   PaStreamParameters captureParameters;
 
   captureParameters.device = recDeviceNum;
   captureParameters.sampleFormat = paFloat32;;
   captureParameters.hostApiSpecificStreamInfo = NULL;
   captureParameters.channelCount = 1;
   if (Pa_GetDeviceInfo(recDeviceNum))
      captureParameters.suggestedLatency =
         Pa_GetDeviceInfo(recDeviceNum)->defaultLowInputLatency;
   else
      captureParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0; 

   // try opening for record and playback
   error = Pa_OpenStream(&stream,
                         &captureParameters, &playbackParameters,
                         highestSampleRate, paFramesPerBufferUnspecified,
                         paClipOff | paDitherOff,
                         audacityAudioCallback, NULL);
   
   if (!error) {
      // Try portmixer for this stream
      mPortMixer = Px_OpenMixer(stream, 0);
      if (!mPortMixer) {
         Pa_CloseStream(stream);
         error = true;
      }
   }

   // if that failed, try just for record
   if( error ) {
      error = Pa_OpenStream(&stream,
                            &captureParameters, NULL,
                            highestSampleRate, paFramesPerBufferUnspecified,
                            paClipOff | paDitherOff,
                            audacityAudioCallback, NULL);

      if (!error) {
         mPortMixer = Px_OpenMixer(stream, 0);
         if (!mPortMixer) {
            Pa_CloseStream(stream);
            error = true;
         }
      }
   }

   // finally, try just for playback
   if ( error ) {
      error = Pa_OpenStream(&stream,
                            NULL, &playbackParameters,
                            highestSampleRate, paFramesPerBufferUnspecified,
                            paClipOff | paDitherOff,
                            audacityAudioCallback, NULL);
      
      if (!error) {
         mPortMixer = Px_OpenMixer(stream, 0);
         if (!mPortMixer) {
            Pa_CloseStream(stream);
            error = true;
         }
      }
   }

   // if it's still not working, give up
   if( error )
      return;

   // Determine mixer capabilities - if it doesn't support control of output
   // signal level, we emulate it (by multiplying this value by all outgoing
   // samples)

   mMixerOutputVol = Px_GetPCMOutputVolume(mPortMixer);
   mEmulateMixerOutputVol = false;
   Px_SetPCMOutputVolume(mPortMixer, 0.0);
   if (Px_GetPCMOutputVolume(mPortMixer) > 0.1)
      mEmulateMixerOutputVol = true;
   Px_SetPCMOutputVolume(mPortMixer, 0.2f);
   if (Px_GetPCMOutputVolume(mPortMixer) < 0.1 ||
       Px_GetPCMOutputVolume(mPortMixer) > 0.3)
      mEmulateMixerOutputVol = true;
   Px_SetPCMOutputVolume(mPortMixer, mMixerOutputVol);

   float inputVol = Px_GetInputVolume(mPortMixer);
   mInputMixerWorks = true;   // assume it works unless proved wrong
   Px_SetInputVolume(mPortMixer, 0.0);
   if (Px_GetInputVolume(mPortMixer) > 0.1)
      mInputMixerWorks = false;  // can't set to zero
   Px_SetInputVolume(mPortMixer, 0.2f);
   if (Px_GetInputVolume(mPortMixer) < 0.1 ||
       Px_GetInputVolume(mPortMixer) > 0.3)
      mInputMixerWorks = false;  // can't set level accurately
   Px_SetInputVolume(mPortMixer, inputVol);

   Pa_CloseStream(stream);

   #if 0
   printf("PortMixer: Output: %s Input: %s\n",
          mEmulateMixerOutputVol? "emulated": "native",
          mInputMixerWorks? "hardware": "no control");
   #endif

   mMixerOutputVol = 1.0;

#endif   // USE_PORTMIXER
}

PaSampleFormat AudacityToPortAudioSampleFormat(sampleFormat format)
{
   switch(format) {
   case int16Sample:
      return paInt16;
   case int24Sample:
      return paInt24;
   case floatSample:
   default:
      return paFloat32;
   }
}

bool AudioIO::StartPortAudioStream(double sampleRate,
                                   unsigned int numPlaybackChannels,
                                   unsigned int numCaptureChannels,
                                   sampleFormat captureFormat)
{
   mLastPaError = paNoError;
   // pick a rate to do the audio I/O at, from those available. The project
   // rate is suggested, but we may get something else if it isn't supported
   mRate = GetBestRate(numCaptureChannels > 0, numPlaybackChannels > 0, sampleRate);
   if (mListener) {
      // advertise the chosen I/O sample rate to the UI
      mListener->OnAudioIORate((int)mRate);
   }
   
   // Special case: Our 24-bit sample format is different from PortAudio's
   // 3-byte packed format. So just make PortAudio return float samples,
   // since we need float values anyway to apply the gain.
   if (captureFormat == int24Sample)
      captureFormat = floatSample;

   mNumPlaybackChannels = numPlaybackChannels;
   mNumCaptureChannels = numCaptureChannels;

   PaStreamParameters *playbackParameters = NULL;
   PaStreamParameters *captureParameters = NULL;
   
   double latencyDuration = DEFAULT_LATENCY_DURATION;
   gPrefs->Read(wxT("/AudioIO/LatencyDuration"), &latencyDuration);

   if( numPlaybackChannels > 0)
   {
      playbackParameters = new PaStreamParameters;
      // this sets the device index to whatever is "right" based on preferences,
      // then defaults
      playbackParameters->device = getPlayDevIndex();
     
      const PaDeviceInfo *playbackDeviceInfo;
      playbackDeviceInfo = Pa_GetDeviceInfo( playbackParameters->device );
      
      if( playbackDeviceInfo == NULL )
         return false;
      
      // regardless of source formats, we always mix to float
      playbackParameters->sampleFormat = paFloat32;
      playbackParameters->hostApiSpecificStreamInfo = NULL;
      playbackParameters->channelCount = mNumPlaybackChannels;

      if (mSoftwarePlaythrough)
         playbackParameters->suggestedLatency =
            playbackDeviceInfo->defaultLowOutputLatency;
      else
         playbackParameters->suggestedLatency = latencyDuration/1000.0;
   }

   if( numCaptureChannels > 0)
   {
      mCaptureFormat = captureFormat;
      
      captureParameters = new PaStreamParameters;
      const PaDeviceInfo *captureDeviceInfo;
      // retrieve the index of the device set in the prefs, or a sensible
      // default if it isn't set/valid
      captureParameters->device = getRecordDevIndex();

      captureDeviceInfo = Pa_GetDeviceInfo( captureParameters->device );

      if( captureDeviceInfo == NULL )
         return false;

      captureParameters->sampleFormat =
         AudacityToPortAudioSampleFormat(mCaptureFormat);

      captureParameters->hostApiSpecificStreamInfo = NULL;
      captureParameters->channelCount = mNumCaptureChannels;

      if (mSoftwarePlaythrough)
         captureParameters->suggestedLatency =
            captureDeviceInfo->defaultHighInputLatency;
      else
         captureParameters->suggestedLatency = latencyDuration/1000.0;
   }

   mLastPaError = Pa_OpenStream( &mPortStreamV19,
                                 captureParameters, playbackParameters,
                                 mRate, paFramesPerBufferUnspecified,
                                 paNoFlag,
                                 audacityAudioCallback, NULL );

#if USE_PORTMIXER
   if (mPortStreamV19 != NULL && mLastPaError == paNoError) {
      #ifdef __WXMAC__
      if (mPortMixer) {
         if (Px_SupportsPlaythrough(mPortMixer)) {
            bool playthrough;

            mPreviousHWPlaythrough = Px_GetPlaythrough(mPortMixer);

            gPrefs->Read(wxT("/AudioIO/Playthrough"), &playthrough, false);
            if (playthrough)
               Px_SetPlaythrough(mPortMixer, 1.0);
            else
               Px_SetPlaythrough(mPortMixer, 0.0);
         }
      }
      #endif
   }
#endif

   // these may be null, but deleting a null pointer should never crash.
   delete captureParameters;
   delete playbackParameters;

   return (mLastPaError == paNoError);
}

void AudioIO::StartMonitoring(double sampleRate)
{
   if ( mPortStreamV19 || mStreamToken )
      return;

   bool success;
   long captureChannels;
   sampleFormat captureFormat = (sampleFormat)
      gPrefs->Read(wxT("/SamplingRate/DefaultProjectSampleFormat"), floatSample);
   gPrefs->Read(wxT("/AudioIO/RecordChannels"), &captureChannels, 2L);
   gPrefs->Read(wxT("/AudioIO/SWPlaythrough"), &mSoftwarePlaythrough, false);
   int playbackChannels = 0;

   if (mSoftwarePlaythrough)
      playbackChannels = 2;

   success = StartPortAudioStream(sampleRate, (unsigned int)playbackChannels,
                                  (unsigned int)captureChannels,
                                  captureFormat);

   // Now start the PortAudio stream!
   mLastPaError = Pa_StartStream( mPortStreamV19 );
}

int AudioIO::StartStream(WaveTrackArray playbackTracks,
                         WaveTrackArray captureTracks,
#ifdef EXPERIMENTAL_MIDI_OUT
                         NoteTrackArray *midiPlaybackTracks,
#endif
                         TimeTrack *timeTrack, double sampleRate,
                         double t0, double t1,
                         AudioIOListener* listener,
                         bool playLooped /* = false */,
                         double cutPreviewGapStart /* = 0.0 */,
                         double cutPreviewGapLen /* = 0.0 */)
{
   if( IsBusy() )
      return 0;

   // We just want to set mStreamToken to -1 - this way avoids
   // an extremely rare but possible race condition, if two functions
   // somehow called StartStream at the same time...
   mStreamToken--;
   if (mStreamToken != -1)
      return 0;

   // TODO: we don't really need to close and reopen stream if the
   // format matches; however it's kind of tricky to keep it open...
   //
   //   if (sampleRate == mRate &&
   //       playbackChannels == mNumPlaybackChannels &&
   //       captureChannels == mNumCaptureChannels &&
   //       captureFormat == mCaptureFormat) {

   if (mPortStreamV19) {
      StopStream();
      while(mPortStreamV19)
         wxMilliSleep( 50 );            
   }

   gPrefs->Read(wxT("/AudioIO/SWPlaythrough"), &mSoftwarePlaythrough, false);
   gPrefs->Read(wxT("/AudioIO/SoundActivatedRecord"), &mPauseRec, false);
   int silenceLevelDB;
   gPrefs->Read(wxT("/AudioIO/SilenceLevel"), &silenceLevelDB, -50);
   int dBRange;
   dBRange = gPrefs->Read(wxT("/GUI/EnvdBRange"), ENV_DB_RANGE);
   if(silenceLevelDB < -dBRange)
   {
      silenceLevelDB = -dBRange + 3;   // meter range was made smaller than SilenceLevel
      gPrefs->Write(wxT("/GUI/EnvdBRange"), dBRange); // so set SilenceLevel reasonable
   }
   mSilenceLevel = (silenceLevelDB + dBRange)/(double)dBRange;  // meter goes -dBRange dB -> 0dB

   mTimeTrack = timeTrack;
   mListener = listener;
   mInputMeter = NULL;
   mOutputMeter = NULL;
   mRate    = sampleRate;
   mT0      = t0;
   mT       = t0;
   mT1      = t1;
   mTime    = t0;
   mSeek    = 0;
   mLastRecordingOffset = 0;
   mPlaybackTracks = playbackTracks;
   mCaptureTracks  = captureTracks;
#ifdef EXPERIMENTAL_MIDI_OUT
   mMidiPlaybackTracks = midiPlaybackTracks;
#endif
   mPlayLooped = playLooped;
   mCutPreviewGapStart = cutPreviewGapStart;
   mCutPreviewGapLen = cutPreviewGapLen;

   double factor = 1.0;
   if (mTimeTrack) {
      factor = mTimeTrack->GetEnvelope()->Average(mT0, mT1);
      factor = (mTimeTrack->GetRangeLower() *
               (1 - factor) +
               factor *
               mTimeTrack->GetRangeUpper()) / 
               100.0;
   }
   mWarpedT1 = factor >= 1 ? mT1 : mT0 + ((mT1 - mT0) / factor);

   //
   // The RingBuffer sizes, and the max amount of the buffer to
   // fill at a time, both grow linearly with the number of
   // tracks.  This allows us to scale up to many tracks without
   // killing performance.
   //

   mPlaybackRingBufferSecs = 4.5 + (0.5 * mPlaybackTracks.GetCount());
   mMaxPlaybackSecsToCopy = 0.75 + (0.25 * mPlaybackTracks.GetCount());

   mCaptureRingBufferSecs = 4.5 + 0.5 * mCaptureTracks.GetCount();   
   mMinCaptureSecsToCopy = 0.2 + (0.2 * mCaptureTracks.GetCount());

   unsigned int playbackChannels = 0;
   unsigned int captureChannels = 0;
   sampleFormat captureFormat = floatSample;

   if( playbackTracks.GetCount() > 0 )
      playbackChannels = 2;

   if (mSoftwarePlaythrough)
      playbackChannels = 2;

   if( captureTracks.GetCount() > 0 )
   {
      // For capture, every input channel gets its own track
      captureChannels = mCaptureTracks.GetCount();
      // I don't deal with the possibility of the capture tracks
      // having different sample formats, since it will never happen
      // with the current code.  This code wouldn't *break* if this
      // assumption was false, but it would be sub-optimal.  For example,
      // if the first track was 16-bit and the second track was 24-bit,
      // we would set the sound card to capture in 16 bits and the second
      // track wouldn't get the benefit of all 24 bits the card is capable
      // of.
      captureFormat = mCaptureTracks[0]->GetSampleFormat();
      
      // Tell project that we are about to start recording
      if (mListener)
         mListener->OnAudioIOStartRecording();
   }

   bool successAudio;

   successAudio = StartPortAudioStream(sampleRate, playbackChannels,
                                       captureChannels, captureFormat);

#ifdef EXPERIMENTAL_MIDI_OUT
   // TODO: it may be that midi out will not work unless audio in or out is
   // active -- this would be a bug and may require a change in the 
   // logic here.

   bool successMidi;

   if(mMidiPlaybackTracks && !mMidiPlaybackTracks->IsEmpty()){
      successMidi = StartPortMidiStream();
   }

   // On the other hand, if MIDI cannot be opened, we will not complain
#endif

   if (!successAudio
#ifdef EXPERIMENTAL_MIDI_OUT
      && !successMidi
#endif
      ) {
      if (mListener && captureChannels > 0)
         mListener->OnAudioIOStopRecording();
      mStreamToken = 0;
      return 0;
   }

   //
   // The (audio) stream has been opened successfully.  We now proceed to 
   // allocate the memory structures the stream will need.
   //

   if( mNumPlaybackChannels > 0 ) {
      // Allocate output buffers.  For every output track we allocate
      // a ring buffer of five seconds
      sampleCount playbackBufferSize =
         (sampleCount)(mRate * mPlaybackRingBufferSecs + 0.5);
      sampleCount playbackMixBufferSize = 
         (sampleCount)(mRate * mMaxPlaybackSecsToCopy + 0.5);
      mPlaybackBuffers = new RingBuffer* [mPlaybackTracks.GetCount()];
      mPlaybackMixers  = new Mixer*      [mPlaybackTracks.GetCount()];

      for( unsigned int i = 0; i < mPlaybackTracks.GetCount(); i++ )
      {
         mPlaybackBuffers[i] = new RingBuffer(floatSample, playbackBufferSize);

         mPlaybackMixers[i]  = new Mixer(1, &mPlaybackTracks[i],
                                         mTimeTrack, mT0, mWarpedT1, 1,
                                         playbackMixBufferSize, false,
                                         mRate, floatSample, false);
         mPlaybackMixers[i]->ApplyTrackGains(false);
      }
   }

   if( mNumCaptureChannels > 0 )
   {
      // Allocate input buffers.  For every input track we allocate
      // a ring buffer of five seconds
      sampleCount captureBufferSize =
         (sampleCount)(mRate * mCaptureRingBufferSecs + 0.5);
      mCaptureBuffers = new RingBuffer* [mCaptureTracks.GetCount()];
      mResample = new Resample* [mCaptureTracks.GetCount()];
      mFactor = sampleRate / mRate;

      for( unsigned int i = 0; i < mCaptureTracks.GetCount(); i++ )
      {
         mCaptureBuffers[i] = new RingBuffer( mCaptureTracks[i]->GetSampleFormat(),
                                              captureBufferSize );
         mResample[i] = new Resample( true, mFactor, mFactor );
      }
   }

#ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
   AILASetStartTime();
#endif

   // We signal the audio thread to call FillBuffers, to prime the RingBuffers
   // so that they will have data in them when the stream starts.  Having the
   // audio thread call FillBuffers here makes the code more predictable, since
   // FillBuffers will ALWAYS get called from the Audio thread.
   mAudioThreadShouldCallFillBuffersOnce = true;

   while( mAudioThreadShouldCallFillBuffersOnce == true )
      wxMilliSleep( 50 );

#ifdef EXPERIMENTAL_MIDI_OUT
   // output MIDI control information up to the current cursor location
   // MOTODO: output midi control changes and program changes up to cursor
   //   QUESTION: can we do that from this thread?
#endif

   if(mNumPlaybackChannels > 0 || mNumCaptureChannels > 0
#ifdef EXPERIMENTAL_MIDI_OUT
      || !mMidiPlaybackTracks->IsEmpty()
#endif
      ) {

      // Now start the PortAudio stream!
      PaError err;
      err = Pa_StartStream( mPortStreamV19 );

      if( err != paNoError )
      {
         // TODO
         // we'll need a more complete way to indicate error.
         // AND we need to delete the ring buffers and mixers, etc.
         if (mListener && mNumCaptureChannels > 0)
            mListener->OnAudioIOStopRecording();
         wxPrintf(wxT("%hs\n"), Pa_GetErrorText(err));
         mStreamToken = 0;
         return 0;
      }
   }

   mAudioThreadFillBuffersLoopRunning = true;

   //
   // Generate an unique value each time, to be returned to
   // clients accessing the AudioIO API, so they can query if
   // are the ones who have reserved AudioIO or not.
   //
   mStreamToken = (++mNextStreamToken);

   return mStreamToken;
}

#ifdef EXPERIMENTAL_MIDI_OUT

#define TIME_PROC ((long (*)(void *)) Pt_Time)

bool AudioIO::IsMidiActive()
{
   if (!mMidiStream) return false;
   // find out if we have passed the last midi event
   return mLastMidiTime + mMidiLatency <= TIME_PROC(NULL);
}


bool AudioIO::StartPortMidiStream() 
{
   int i, latency; 
   
   // Only start MIDI stream if there is an open track
   if (mMidiPlaybackTracks->GetCount() == 0)
      return false;

   /* get latency from PortAudio */
   int framesPerBuffer = 1102; // constant passed to Pa_OpenStream call 
                               // but not defined beforehand
/* HCK MIDI FIX ORG
   int numBuffers = Pa_GetMinNumBuffers( framesPerBuffer, mRate );

   if (numBuffers)
      latency = 1000 * numBuffers * framesPerBuffer / mRate;
   else
      latency = 500;
HCK MIDI PATCH ORG */

   mMidiLatency = 100;
   printf("StartPortMidiStream: mT0 %g mTime %g\n", 
          gAudioIO->mT0, gAudioIO->mTime);
   if (Pt_Started()) Pt_Stop(); // start from zero
   Pt_Start(1, 0, 0); /* timer started w/millisecond accuracy */

   /* get midi playback device */
   PmDeviceID playbackDevice = Pm_GetDefaultOutputDeviceID();
   wxString playbackDeviceName = gPrefs->Read(wxT("/MidiIO/PlaybackDevice"), 
                                              wxT(""));

   if (wxStrcmp(playbackDeviceName, wxT("")) != 0) {
      for (i = 0; i < Pm_CountDevices(); i++) {
         const PmDeviceInfo *info = Pm_GetDeviceInfo(i);
         if (!info) continue;
         wxString interf(info->interf, wxConvLocal);
         wxString name(info->name, wxConvLocal);
         wxString device = wxString::Format(wxT("%s: %s"),
                                            interf.c_str(),
                                            name.c_str());
        if (wxStrcmp(device, playbackDeviceName) == 0) {
            playbackDevice = i;
         }
      }
   } // (else playback device has Pm_GetDefaultOuputDeviceID())

   /* open output device */
   mLastPmError = Pm_OpenOutput(&mMidiStream, 
                                playbackDevice, 
                                NULL,
                                0, 
                                TIME_PROC,
                                NULL, 
                                mMidiLatency);
   // DEBUGGING
      const PmDeviceInfo *info = Pm_GetDeviceInfo(playbackDevice);
      printf("Pm_OpenOutput on %s, return code %d\n", 
             info->name, mLastPmError);

   mCurrentMidiTime = TIME_PROC(NULL);
   mLastMidiTime = 0x7FFF0000; // a big number, adjusted when we play last
                               // midi event
   mMidiWait = 0;

   fprintf(stderr, "mT0: %f\n", mT0);
   fprintf(stderr, "%li %li : STARTING\n", mCurrentMidiTime, mLastMidiTime );

   mMidiStreamActive = true;
   //   mCnt = 0;

   // for now, play only one track
   mSeq = (*mMidiPlaybackTracks)[0]->GetSequence();
   mVC = (*mMidiPlaybackTracks)[0]->GetVisibleChannels();
 
   mIterator = new Alg_iterator(mSeq, true);
   mIterator->begin();
   GetNextEvent(); // prime the pump for FillMidiBuffers


   // Start MIDI from current cursor position
   mSendMidiState = true;
   while (mNextEvent && 
          mNextEvent->time < mT0) {
      OutputEvent();
      GetNextEvent();
   }
   // MOTODO -- fix this to send updates immediately until t0 is reached
   mSendMidiState = false;

   return (mLastPmError == pmNoError);      
}
#endif

void AudioIO::SetMeters(Meter *inputMeter, Meter *outputMeter)
{
   mInputMeter = inputMeter;
   mOutputMeter = outputMeter;

   if (mInputMeter)
      mInputMeter->Reset(mRate, true);
   if (mOutputMeter)
      mOutputMeter->Reset(mRate, true);

   mUpdateMeters = true;
}

void AudioIO::StopStream()
{
   if( mPortStreamV19 == NULL 
#ifdef EXPERIMENTAL_MIDI_OUT
       && mMidiStream == NULL 
#endif
     )
      return;

   if( Pa_IsStreamStopped( mPortStreamV19 ) 
#ifdef EXPERIMENTAL_MIDI_OUT
       && !mMidiStreamActive 
#endif
     )
      return;

   // Avoid race condition by making sure this function only
   // gets called once at a time
   mStopStreamCount++; // <- note that this is not atomic, therefore has
                       // a race condition -RBD
   if (mStopStreamCount != 1)
      return;

   //
   // We got here in one of two ways:
   //
   // 1. The user clicked the stop button and we therefore want to stop
   //    as quickly as possible.  So we use AbortStream().  If this is
   //    the case the portaudio stream is still in the Running state
   //    (see PortAudio state machine docs).
   //
   // 2. The callback told PortAudio to stop the stream since it had
   //    reached the end of the selection.  The UI thread discovered
   //    this by noticing that AudioIO::IsActive() returned false.
   //    IsActive() (which calls Pa_GetStreamActive()) will not return
   //    false until all buffers have finished playing, so we can call
   //    AbortStream without losing any samples.  If this is the case
   //    we are in the "callback finished state" (see PortAudio state
   //    machine docs).
   //
   // The moral of the story: We can call AbortStream safely, without
   // losing samples.
   //
   // DMM: This doesn't seem to be true; it seems to be necessary to
   // call StopStream if the callback brought us here, and AbortStream
   // if the user brought us here.
   //

   mAudioThreadFillBuffersLoopRunning = false;

   // Audacity can deadlock if it tries to update meters while
   // we're stopping PortAudio (because the meter updating code
   // tries to grab a UI mutex while PortAudio tries to join a
   // pthread).  So we tell the callback to stop updating meters,
   // and wait until the callback has left this part of the code
   // if it was already there.
   mUpdateMeters = false;
   while(mUpdatingMeters) {
      wxYield();
      wxMilliSleep( 50 );
   }

   // Turn off HW playthrough if PortMixer is being used

  #if defined(USE_PORTMIXER)
   if( mPortMixer ) {
      #if __WXMAC__
      if (Px_SupportsPlaythrough(mPortMixer) && mPreviousHWPlaythrough >= 0.0)
         Px_SetPlaythrough(mPortMixer, mPreviousHWPlaythrough);
         mPreviousHWPlaythrough = -1.0;
      #endif
   }
  #endif

   if (mPortStreamV19) {
      Pa_AbortStream( mPortStreamV19 );
      Pa_CloseStream( mPortStreamV19 );
      mPortStreamV19 = NULL;
   }


#ifdef EXPERIMENTAL_MIDI_OUT
   /* Stop Midi playback */
   // problem here -- if explicitly stopped, we'll hang here
   //   while (mLastMidiTime + mMidiLatency + 10 < TIME_PROC(NULL)) {
   //      wxMilliSleep( 50 );
   //   }
      
   if ( mMidiStream ) {
      mMidiStreamActive = false;
      // MOTODO: if output in progress, send all off, etc.
      for (int i = 0; i < 16; i++) {
         Pm_WriteShort(mMidiStream, 0, Pm_Message(0xB0 + i, 0x7B, 0));
      }
      wxMilliSleep(40); // deliver the all-off messages before closing
      Pm_Abort(mMidiStream); // MOTODO: only abort if output in progress
      Pm_Close(mMidiStream);
      printf("Pm_Close() called\n");
      mMidiStream = NULL;

      // Reset MIDI track positions this way for now
      // mMidiPlaybackTracks[0]->SetLastMidiPosition(0);
      // mLastMidiTime = 0;
      mIterator->end();
   }
#endif

   // If there's no token, we were just monitoring, so we can
   // skip this next part...
   if (mStreamToken > 0) {
      // In either of the above cases, we want to make sure that any
      // capture data that made it into the PortAudio callback makes it
      // to the target WaveTrack.  To do this, we ask the audio thread to
      // call FillBuffers one last time (it normally would not do so since
      // Pa_GetStreamActive() would now return false
      mAudioThreadShouldCallFillBuffersOnce = true;

      while( mAudioThreadShouldCallFillBuffersOnce == true )
      {
         // LLL:  Experienced recursive yield here...once.
         wxGetApp().Yield( true );
         wxMilliSleep( 50 );
      }

      //
      // Everything is taken care of.  Now, just free all the resources
      // we allocated in StartStream()
      //
      
      if( mPlaybackTracks.GetCount() > 0 )
      {
         for( unsigned int i = 0; i < mPlaybackTracks.GetCount(); i++ )
         {
            delete mPlaybackBuffers[i];
            delete mPlaybackMixers[i];
         }
         
         delete[] mPlaybackBuffers;
         delete[] mPlaybackMixers;
      }

      //
      // Offset all recorded tracks to account for latency
      //
      if( mCaptureTracks.GetCount() > 0 )
      {
         //
         // We only apply latency correction when we actually played back
         // tracks during the recording. If we did not play back tracks,
         // there's nothing we could be out of sync with. This also covers the
         // case that we do not apply latency correction when recording the
         // first track in a project.
         //
         double latencyCorrection = DEFAULT_LATENCY_CORRECTION;
         gPrefs->Read(wxT("/AudioIO/LatencyCorrection"), &latencyCorrection);
         
         double recordingOffset =
            mLastRecordingOffset + latencyCorrection / 1000.0;

         for( unsigned int i = 0; i < mCaptureTracks.GetCount(); i++ )
            {
               delete mCaptureBuffers[i];
               delete mResample[i];
               
               WaveTrack* track = mCaptureTracks[i];
               track->Flush();

               if (mPlaybackTracks.GetCount() > 0)
               {  // only do latency correction if some tracks are being played back
                  WaveTrackArray playbackTracks;
                  AudacityProject *p = GetActiveProject();
                  // we need to get this as mPlaybackTracks does not contain tracks being recorded into
                  playbackTracks = p->GetTracks()->GetWaveTrackArray(false);
                  bool appendRecord = false;
                  for( unsigned int j = 0; j < playbackTracks.GetCount(); j++)
                  {  // find if we are recording into an existing track (append-record)
                     WaveTrack* trackP = playbackTracks[j];
                     if( track == trackP )
                     {
                        if( track->GetStartTime() != mT0 )  // in a new track if these are equal
                        {
                           appendRecord = true;
                           break;
                        }
                     }
                  }
                  if( appendRecord )
                  {  // append-recording
                     if (recordingOffset < 0)
                        track->Clear(mT0, mT0 - recordingOffset); // cut the latency out
                     else
                        track->InsertSilence(mT0, recordingOffset); // put silence in
                  }
                  else
                  {  // recording into a new track
                     track->SetOffset(track->GetStartTime() + recordingOffset);
                     if(track->GetEndTime() < 0.)
                     {
                        wxMessageDialog m(NULL, _("Latency Correction setting has caused the recorded audio to be hidden before zero.\nAudacity has brought it back to start at zero.\nYou may have to use the Time Shift Tool (<---> or F5) to drag the track to the right place."),
                           _("Latency problem"), wxOK);
                        m.ShowModal();
                        track->SetOffset(0.);
                     }
                  }
               }
            }
         
         delete[] mCaptureBuffers;
         delete[] mResample;
      }
   }

   if (mInputMeter)
      mInputMeter->Reset(mRate, false);
   if (mOutputMeter)
      mOutputMeter->Reset(mRate, false);

   if (mListener && mNumCaptureChannels > 0)
      mListener->OnAudioIOStopRecording();
      
   //
   // Only set token to 0 after we're totally finished with everything
   //
   mStreamToken = 0;
   mStopStreamCount = 0;
}

void AudioIO::SetPaused(bool state)
{
   mPaused = state;
}

bool AudioIO::IsPaused()
{
   return mPaused;
}

bool AudioIO::IsBusy()
{
   if (mStreamToken != 0)
      return true;

   return false;
}

bool AudioIO::IsStreamActive()
{
   bool isActive = false;
   if( mPortStreamV19 )
      isActive = (Pa_IsStreamActive( mPortStreamV19 ) > 0);
   else isActive = false;

/* REQUIRES PORTMIDI */
//   if( mMidiStreamActive )
//      isActive = true;
   return isActive;
}

bool AudioIO::IsStreamActive(int token)
{
   if( IsStreamActive() && token > 0 && token == mStreamToken )
      return true;
   else
      return false;
}

bool AudioIO::IsAudioTokenActive(int token)
{
   return ( token > 0 && token == mStreamToken );
}

bool AudioIO::IsMonitoring()
{
   return ( mPortStreamV19 && mStreamToken==0 );
}

double AudioIO::NormalizeStreamTime(double absoluteTime) const
{
   // dmazzoni: This function is needed for two reasons:
   // One is for looped-play mode - this function makes sure that the
   // position indicator keeps wrapping around.  The other reason is
   // more subtle - it's because PortAudio can query the hardware for
   // the current stream time, and this query is not always accurate.
   // Sometimes it's a little behind or ahead, and so this function
   // makes sure that at least we clip it to the selection.
   //
   // msmeyer: There is also the possibility that we are using "cut preview"
   //          mode. In this case, we should jump over a defined "gap" in the
   //          audio.

   // msmeyer: Just to be sure, the returned stream time should
   //          never be smaller than the actual start time.
   if (absoluteTime < mT0)
      absoluteTime = mT0;

   if (absoluteTime > mT1)
      absoluteTime = mT1;
   
   if (mCutPreviewGapLen > 0)
   {
      // msmeyer: We're in cut preview mode, so if we are on the right
      // side of the gap, we jump over it.
      if (absoluteTime > mCutPreviewGapStart)
         absoluteTime += mCutPreviewGapLen;
   }

   return absoluteTime;
}

double AudioIO::GetStreamTime()
{
   if( !IsStreamActive() )
      return -1000000000;

   return NormalizeStreamTime(mTime);
}


wxArrayLong AudioIO::GetSupportedPlaybackRates(int devIndex, double rate)
{
   if (devIndex == -1)
   {  // weren't given a device index, get the prefs / default one
      devIndex = getPlayDevIndex();
   }

   // Check if we can use the cached rates
   if (mCachedPlaybackIndex != -1 && devIndex == mCachedPlaybackIndex
         && rate == 0.0)
   {
      return mCachedPlaybackRates;
   }

   wxArrayLong supported;
   int irate = (int)rate;
   const PaDeviceInfo* devInfo = NULL;
   int i;

   wxLogDebug(wxT("Getting supported playback rates for device %d"), devIndex);
   devInfo = Pa_GetDeviceInfo(devIndex);
   
   if (!devInfo)
   {
      wxLogDebug(wxT("GetSupportedPlaybackRates() Could not get device info!"));
      return supported;
   }

   PaStreamParameters pars;

   pars.device = devIndex;
   pars.channelCount = 1;
   pars.sampleFormat = paFloat32;
   pars.suggestedLatency = devInfo->defaultHighOutputLatency;
   pars.hostApiSpecificStreamInfo = NULL;
   
   for (i = 0; i < NumRatesToTry; i++)
   {
      if (Pa_IsFormatSupported(NULL, &pars, RatesToTry[i]) == 0)
      {
         wxLogDebug(wxT("Rate %ld Hz is supported"), RatesToTry[i]);
         supported.Add(RatesToTry[i]);
      }
   }

   if (irate != 0 && supported.Index(irate) == wxNOT_FOUND)
   {
      if (Pa_IsFormatSupported(NULL, &pars, irate) == 0)
      {
         wxLogDebug(wxT("Suggested rate %ld Hz is supported"), irate);
         supported.Add(irate);
      }
   }

   return supported;
}

wxArrayLong AudioIO::GetSupportedCaptureRates(int devIndex, double rate)
{
   if (devIndex == -1)
   {  // not given a device, look up in prefs / default
      devIndex = getRecordDevIndex();
   }

   // Check if we can use the cached rates
   if (mCachedCaptureIndex != -1 && devIndex == mCachedCaptureIndex
         && rate == 0.0)
   {
      return mCachedCaptureRates;
   }

   wxArrayLong supported;
   int irate = (int)rate;
   const PaDeviceInfo* devInfo = NULL;
   int i;

   wxLogDebug(wxT("Getting supported capture rates for device %d"), devIndex);
   devInfo = Pa_GetDeviceInfo(devIndex);

   if (!devInfo)
   {
      wxLogDebug(wxT("GetSupportedCaptureRates() Could not get device info!"));
      return supported;
   }

   double latencyDuration = DEFAULT_LATENCY_DURATION;
   long recordChannels = 1;
   gPrefs->Read(wxT("/AudioIO/LatencyDuration"), &latencyDuration);
   gPrefs->Read(wxT("/AudioIO/RecordChannels"), &recordChannels);

   PaStreamParameters pars;

   pars.device = devIndex;
   pars.channelCount = recordChannels;
   pars.sampleFormat = paFloat32;
   pars.suggestedLatency = latencyDuration / 1000.0;
   pars.hostApiSpecificStreamInfo = NULL;
   
   for (i = 0; i < NumRatesToTry; i++)
   {
      if (Pa_IsFormatSupported(&pars, NULL, RatesToTry[i]) == 0)
      {
         wxLogDebug(wxT("Rate %ld Hz is supported"), RatesToTry[i]);
         supported.Add(RatesToTry[i]);
      }
   }

   if (irate != 0 && supported.Index(irate) == wxNOT_FOUND)
   {
      if (Pa_IsFormatSupported(&pars, NULL, irate) == 0)
      {
         wxLogDebug(wxT("Suggested rate %ld Hz is supported"), irate);
         supported.Add(irate);
      }
   }

   return supported;
}

wxArrayLong AudioIO::GetSupportedSampleRates(int playDevice, int recDevice, double rate)
{
   // Not given device indices, look up prefs
   if (playDevice == -1) {
      playDevice = getPlayDevIndex();
   }
   if (recDevice == -1) {
      recDevice = getRecordDevIndex();
   }

   // Check if we can use the cached rates
   if (mCachedPlaybackIndex != -1 && mCachedCaptureIndex != -1 && 
         playDevice == mCachedPlaybackIndex &&
         recDevice == mCachedCaptureIndex &&
         rate == 0.0)
   {
      return mCachedSampleRates;
   }

   wxArrayLong playback = GetSupportedPlaybackRates(playDevice, rate);
   wxArrayLong capture = GetSupportedCaptureRates(recDevice, rate);
   int i;

   // Return only sample rates which are in both arrays
   wxArrayLong result;

   for (i = 0; i < (int)playback.GetCount(); i++)
      if (capture.Index(playback[i]) != wxNOT_FOUND)
         result.Add(playback[i]);

   // If this yields no results, use the default sample rates nevertheless
/*   if (result.IsEmpty())
   {
      for (i = 0; i < NumStandardRates; i++)
         result.Add(StandardRates[i]);
   }*/

   return result;
}

/** \todo: should this take into account PortAudio's value for 
 * PaDeviceInfo::defaultSampleRate? In principal this should let us work out
 * which rates are "real" and which resampled in the drivers, and so prefer
 * the real rates. */
int AudioIO::GetOptimalSupportedSampleRate()
{
   wxArrayLong rates = GetSupportedSampleRates();

   if (rates.Index(44100) != wxNOT_FOUND)
      return 44100;

   if (rates.Index(48000) != wxNOT_FOUND)
      return 48000;

   // if there are no supported rates, the next bit crashes. So check first,
   // and give them a "sensible" value if there are no valid values. They
   // will still get an error later, but with any luck may have changed
   // something by then. It's no worse than having an invalid default rate
   // stored in the preferences, which we don't check for
   if (rates.IsEmpty()) return 44100;

   return rates[rates.GetCount() - 1];
}

double AudioIO::GetBestRate(bool capturing, bool playing, double sampleRate)
{
   // Check if we can use the cached value
   if (mCachedBestRateIn != 0.0 && mCachedBestRateIn == sampleRate) {
      return mCachedBestRateOut;
   }

   // In order to cache the value, all early returns should instead set retval
   // and jump to finished
   double retval;

   wxArrayLong rates;
   if (capturing) wxLogDebug(wxT("AudioIO::GetBestRate() for capture"));
   if (playing) wxLogDebug(wxT("AudioIO::GetBestRate() for playback"));
   wxLogDebug(wxT("GetBestRate() suggested rate %.0lf Hz"), sampleRate);

   if (capturing && !playing) {
      rates = GetSupportedCaptureRates(-1, sampleRate);
   }
   else if (playing && !capturing) {
      rates = GetSupportedPlaybackRates(-1, sampleRate);
   }
   else {   // we assume capturing and playing - the alternative would be a 
            // bit odd
      rates = GetSupportedSampleRates(-1, -1, sampleRate);
   }
   /* rem rates is the array of hardware-supported sample rates (in the current
    * configuration), sampleRate is the Project Rate (desired sample rate) */
   long rate = (long)sampleRate;
   
   if (rates.Index(rate) != wxNOT_FOUND) {
      wxLogDebug(wxT("GetBestRate() Returning %.0ld Hz"), rate);
      retval = rate;
      goto finished;
      /* the easy case - the suggested rate (project rate) is in the list, and
       * we can just accept that and send back to the caller. This should be
       * the case for most users most of the time (all of the time on
       * Win MME as the OS does resampling) */
   }

   /* if we get here, there is a problem - the project rate isn't supported
    * on our hardware, so we can't us it. Need to come up with an alternative
    * rate to use. The process goes like this:
    * * If there are no rates to pick from, we're stuck and return 0 (error)
    * * If there are some rates, we pick the next one higher than the requested
    *   rate to use.
    * * If there aren't any higher, we use the highest available rate */

   if (rates.IsEmpty()) {
      /* we're stuck - there are no supported rates with this hardware. Error */
      wxLogDebug(wxT("GetBestRate() Error - no supported sample rates"));
      retval = 0.0;
      goto finished;
   }
   int i;
   for (i = 0; i < (int)rates.GetCount(); i++)  // for each supported rate
         {
         if (rates[i] > rate) {
            // supported rate is greater than requested rate
            wxLogDebug(wxT("GetBestRate() Returning next higher rate - %.0ld Hz"), rates[i]);
            retval = rates[i];
            goto finished;
         }
         }

   wxLogDebug(wxT("GetBestRate() Returning highest rate - %.0ld Hz"), rates[rates.GetCount() - 1]);
   retval = rates[rates.GetCount() - 1]; // the highest available rate
   goto finished;

finished:
   mCachedBestRateIn = sampleRate;
   mCachedBestRateOut = retval;
   return retval;
}      


//////////////////////////////////////////////////////////////////////
//
//     Audio Thread Context
//
//////////////////////////////////////////////////////////////////////

AudioThread::ExitCode AudioThread::Entry()
{
   while( !TestDestroy() )
   {
      // Set LoopActive outside the tests to avoid race condition
      gAudioIO->mAudioThreadFillBuffersLoopActive = true;
      if( gAudioIO->mAudioThreadShouldCallFillBuffersOnce )
      {
         gAudioIO->FillBuffers();
         gAudioIO->mAudioThreadShouldCallFillBuffersOnce = false;
      }
      else if( gAudioIO->mAudioThreadFillBuffersLoopRunning )
      {
         gAudioIO->FillBuffers();
      }
      gAudioIO->mAudioThreadFillBuffersLoopActive = false;

#ifdef EXPERIMENTAL_MIDI_OUT
     if( gAudioIO->mMidiStreamActive && 
         gAudioIO->mAudioThreadFillBuffersLoopRunning)
      {
         gAudioIO->FillMidiBuffers();
      }
#endif
      Sleep(10);
   }

   return 0;
}

int AudioIO::GetCommonlyAvailPlayback()
{
   int commonlyAvail = mPlaybackBuffers[0]->AvailForPut();
   unsigned int i;

   for( i = 1; i < mPlaybackTracks.GetCount(); i++ )
   {
      int thisBlockAvail = mPlaybackBuffers[i]->AvailForPut();

      if( thisBlockAvail < commonlyAvail )
         commonlyAvail = thisBlockAvail;
   }

   return commonlyAvail;
}

int AudioIO::GetCommonlyAvailCapture()
{
   int commonlyAvail = mCaptureBuffers[0]->AvailForGet();
   unsigned int i;

   for( i = 1; i < mCaptureTracks.GetCount(); i++ )
   {
      int avail = mCaptureBuffers[i]->AvailForGet();
      if( avail < commonlyAvail )
         commonlyAvail = avail;
   }

   return commonlyAvail;
}

int AudioIO::getRecordDevIndex(wxString devName)
{
   // if we don't get given a device, look up the preferences
   if (devName.IsEmpty())
   {
      devName = gPrefs->Read(wxT("/AudioIO/RecordingDevice"), wxT(""));
   }

   int i;
   for (i = 0; i < Pa_GetDeviceCount(); i++)
   {
      const PaDeviceInfo* info = Pa_GetDeviceInfo(i);

      if (info && (DeviceName(info) == devName) && (info->maxInputChannels > 0))
      {
         // this device name matches the stored one, and works.
         // So we say this is the answer and return it
         return i;
      }
   }
   // landing here, we either don't have a value in the preferences, or 
   // the stored / supplied value doesn't exist on the system. So we need to
   // use a default value
   int recDeviceNum = Pa_GetDefaultInputDevice();

   // Sometimes PortAudio returns -1 if it cannot find a suitable default
   // device, so we just use the first one available
   if (recDeviceNum < 0)
      recDeviceNum = 0;
   return recDeviceNum;
}

int AudioIO::getPlayDevIndex(wxString devName )
{
   // if we don't get given a device, look up the preferences
   if (devName.IsEmpty())
   {
      devName = gPrefs->Read(wxT("/AudioIO/PlaybackDevice"), wxT(""));
   }

   int i;
   for (i = 0; i < Pa_GetDeviceCount(); i++)
   {
      const PaDeviceInfo* info = Pa_GetDeviceInfo(i);

      if (info && (DeviceName(info) == devName) && (info->maxOutputChannels > 0))
      {
         // this device name matches the stored one, and works.
         // So we say this is the answer and return it
         return i;
      }
   }
   // landing here, we either don't have a value in the preferences, or 
   // the stored / supplied value doesn't exist on the system. So we need to
   // use a default value
   int DeviceNum = Pa_GetDefaultOutputDevice();
   // Sometimes PortAudio returns -1 if it cannot find a suitable default
   // device, so we just use the first one available
   if (DeviceNum < 0)
      DeviceNum = 0;
   return DeviceNum;
}

wxString AudioIO::GetDeviceInfo()
{
   wxStringOutputStream o;
   wxTextOutputStream s(o, wxEOL_UNIX);
   wxString e(wxT("\n"));

   if (IsStreamActive()) {
      return wxT("Stream is active ... unable to gather information.");
   }


   int recDeviceNum = Pa_GetDefaultInputDevice();
   int playDeviceNum = Pa_GetDefaultOutputDevice();

   int cnt = Pa_GetDeviceCount();

   wxLogDebug(wxT("Portaudio reports %d audio devices"),cnt);

   s << wxT("==============================") << e;
   s << wxT("Default capture device number: ") << recDeviceNum << e;
   s << wxT("Default playback device number: ") << playDeviceNum << e;

   wxString recDevice = gPrefs->Read(wxT("/AudioIO/RecordingDevice"), wxT(""));
   wxString playDevice = gPrefs->Read(wxT("/AudioIO/PlaybackDevice"), wxT(""));
   int j;

   // This gets info on all available audio devices (input and output)
   if (cnt <= 0) {
      s << wxT("No devices found\n");
      return o.GetString();
   }

   const PaDeviceInfo* info;
     
   for (j = 0; j < cnt; j++) {
      s << wxT("==============================") << e;

      info = Pa_GetDeviceInfo(j);
      if (!info) {
         s << wxT("Device info unavailable for: ") << j << wxT("\n");
         continue;
      }

      wxString name = DeviceName(info);

      s << wxT("Device ID: ") << j << e;
      s << wxT("Device name: ") << name << e;
      s << wxT("Input channels: ") << info->maxInputChannels << e;
      s << wxT("Output channels: ") << info->maxOutputChannels << e;
      s << wxT("Low Input Latency: ") << info->defaultLowInputLatency << e;
      s << wxT("Low Output Latency: ") << info->defaultLowOutputLatency << e;
      s << wxT("High Input Latency: ") << info->defaultHighInputLatency << e;
      s << wxT("High Output Latency: ") << info->defaultHighOutputLatency << e;

      wxArrayLong rates = GetSupportedPlaybackRates(j, 0.0);

      s << wxT("Supported Rates:") << e;
      for (int k = 0; k < (int) rates.GetCount(); k++) {
         s << wxT("    ") << (int)rates[k] << e;
      }

      if (name == playDevice && info->maxOutputChannels > 0)
         playDeviceNum = j;

      if (name == recDevice && info->maxInputChannels > 0)
         recDeviceNum = j;

      // Sometimes PortAudio returns -1 if it cannot find a suitable default
      // device, so we just use the first one available
      if (recDeviceNum < 0 && info->maxInputChannels > 0){
         recDeviceNum = j;
      }
      if (playDeviceNum < 0 && info->maxOutputChannels > 0){
         playDeviceNum = j;
      }
   }

   bool haveRecDevice = (recDeviceNum >= 0);
   bool havePlayDevice = (playDeviceNum >= 0);

   s << wxT("==============================") << e;
   if(haveRecDevice){
      s << wxT("Selected capture device: ") << recDeviceNum << wxT(" - ") << recDevice << e;
   }else{
      s << wxT("No capture device found.") << e;
   }
   if(havePlayDevice){
      s << wxT("Selected playback device: ") << playDeviceNum << wxT(" - ") << playDevice << e;
   }else{
      s << wxT("No playback device found.") << e;
   }   

   wxArrayLong supportedSampleRates;

   if(havePlayDevice && haveRecDevice){
      supportedSampleRates = GetSupportedSampleRates(playDeviceNum, recDeviceNum);

      s << wxT("Supported Rates:") << e;
      for (int k = 0; k < (int) supportedSampleRates.GetCount(); k++) {
         s << wxT("    ") << (int)supportedSampleRates[k] << e;
      }
   }else{
      s << wxT("Cannot check mutual sample rates without both devices.") << e;
      return o.GetString();
   }

#if defined(USE_PORTMIXER)
   if (supportedSampleRates.GetCount() > 0)
      {
      int highestSampleRate = supportedSampleRates[supportedSampleRates.GetCount() - 1];
      bool EmulateMixerInputVol = true;
      bool EmulateMixerOutputVol = true;
      float MixerInputVol = 1.0;
      float MixerOutputVol = 1.0;

      int error;

      PaStream *stream;
   
      PaStreamParameters playbackParameters;

      playbackParameters.device = playDeviceNum;
      playbackParameters.sampleFormat = paFloat32;
      playbackParameters.hostApiSpecificStreamInfo = NULL;
      playbackParameters.channelCount = 1;
      if (Pa_GetDeviceInfo(playDeviceNum)){
         playbackParameters.suggestedLatency =
            Pa_GetDeviceInfo(playDeviceNum)->defaultLowOutputLatency;
      }
      else{
         playbackParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0; 
      }

      PaStreamParameters captureParameters;
 
      captureParameters.device = recDeviceNum;
      captureParameters.sampleFormat = paFloat32;;
      captureParameters.hostApiSpecificStreamInfo = NULL;
      captureParameters.channelCount = 1;
      if (Pa_GetDeviceInfo(recDeviceNum)){
         captureParameters.suggestedLatency =
            Pa_GetDeviceInfo(recDeviceNum)->defaultLowInputLatency;
      }else{
         captureParameters.suggestedLatency = DEFAULT_LATENCY_CORRECTION/1000.0; 
      }

      error = Pa_OpenStream(&stream,
                         &captureParameters, &playbackParameters,
                         highestSampleRate, paFramesPerBufferUnspecified,
                         paClipOff | paDitherOff,
                         audacityAudioCallback, NULL);

      if (error) {
         error = Pa_OpenStream(&stream,
                            &captureParameters, NULL,
                            highestSampleRate, paFramesPerBufferUnspecified,
                            paClipOff | paDitherOff,
                            audacityAudioCallback, NULL);
      }

      if (error) {
         s << wxT("Recieved ") << error << wxT(" while opening devices") << e;
         return o.GetString();
      }

      PxMixer *PortMixer = Px_OpenMixer(stream, 0);

      if (!PortMixer) {
         s << wxT("Unable to open Portmixer") << e;
         Pa_CloseStream(stream);
         return o.GetString();
      }

      s << wxT("==============================") << e;
      s << wxT("Available mixers:") << e;

      cnt = Px_GetNumMixers(stream);
      for (int i = 0; i < cnt; i++) {
         wxString name(Px_GetMixerName(stream, i), wxConvLocal);
         s << i << wxT(" - ") << name << e;
      }

      s << wxT("==============================") << e;
      s << wxT("Available capture sources:") << e;
      cnt = Px_GetNumInputSources(PortMixer);
      for (int i = 0; i < cnt; i++) {
         wxString name(Px_GetInputSourceName(PortMixer, i), wxConvLocal);
         s << i << wxT(" - ") << name << e;
      }

      s << wxT("==============================") << e;
      s << wxT("Available playback volumes:") << e;
      cnt = Px_GetNumOutputVolumes(PortMixer);
      for (int i = 0; i < cnt; i++) {
         wxString name(Px_GetOutputVolumeName(PortMixer, i), wxConvLocal);
         s << i << wxT(" - ") << name << e;
      }

      // Determine mixer capabilities - it it doesn't support either
      // input or output, we emulate them (by multiplying this value
      // by all incoming/outgoing samples)

      MixerOutputVol = Px_GetPCMOutputVolume(PortMixer);
      EmulateMixerOutputVol = false;
      Px_SetPCMOutputVolume(PortMixer, 0.0);
      if (Px_GetPCMOutputVolume(PortMixer) > 0.1)
         EmulateMixerOutputVol = true;
      Px_SetPCMOutputVolume(PortMixer, 0.2f);
      if (Px_GetPCMOutputVolume(PortMixer) < 0.1 ||
          Px_GetPCMOutputVolume(PortMixer) > 0.3)
         EmulateMixerOutputVol = true;
      Px_SetPCMOutputVolume(PortMixer, MixerOutputVol);

      MixerInputVol = Px_GetInputVolume(PortMixer);
      EmulateMixerInputVol = false;
      Px_SetInputVolume(PortMixer, 0.0);
      if (Px_GetInputVolume(PortMixer) > 0.1)
         EmulateMixerInputVol = true;
      Px_SetInputVolume(PortMixer, 0.2f);
      if (Px_GetInputVolume(PortMixer) < 0.1 ||
          Px_GetInputVolume(PortMixer) > 0.3)
         EmulateMixerInputVol = true;
      Px_SetInputVolume(PortMixer, MixerInputVol);
   
      Pa_CloseStream(stream);
   
      s << wxT("==============================") << e;
      s << wxT("Capture volume is ") << (EmulateMixerInputVol? wxT("emulated"): wxT("native")) << e;
      s << wxT("Playback volume is ") << (EmulateMixerOutputVol? wxT("emulated"): wxT("native")) << e;
   
      Px_CloseMixer(PortMixer);

      }  //end of massive if statement if a valid sample rate has been found
#endif
   return o.GetString();
}

// This method is the data gateway between the audio thread (which
// communicates with the disk) and the PortAudio callback thread
// (which communicates with the audio device).
void AudioIO::FillBuffers()
{
   unsigned int i;

   if( mPlaybackTracks.GetCount() > 0 )
   {
      // Though extremely unlikely, it is possible that some buffers
      // will have more samples available than others.  This could happen
      // if we hit this code during the PortAudio callback.  To keep
      // things simple, we only write as much data as is vacant in
      // ALL buffers, and advance the global time by that much.
      int commonlyAvail = GetCommonlyAvailPlayback();

      //
      // Determine how much this will globally advance playback time
      //
      double secsAvail = commonlyAvail / mRate;

      //
      // Don't fill the buffers at all unless we can do the
      // full mMaxPlaybackSecsToCopy.  This improves performance
      // by not always trying to process tiny chunks, eating the
      // CPU unnecessarily.
      //
      // The exception is if we're at the end of the selected
      // region - then we should just fill the buffer.
      //
      if (secsAvail >= mMaxPlaybackSecsToCopy ||
          (!mPlayLooped && (secsAvail > 0 && mT+secsAvail >= mWarpedT1)))
      {
         // Limit maximum buffer size (increases performance)
         if (secsAvail > mMaxPlaybackSecsToCopy)
            secsAvail = mMaxPlaybackSecsToCopy;

         double deltat;

         // msmeyer: When playing a very short selection in looped
         // mode, the selection must be copied to the buffer multiple
         // times, to ensure, that the buffer has a reasonable size
         // This is the purpose of this loop.
         do {
            deltat = secsAvail;
            if( mT + deltat > mWarpedT1 )
            {
               deltat = mWarpedT1 - mT;
               if( deltat < 0.0 )
                  deltat = 0.0;
            }
            mT += deltat;
            
            secsAvail -= deltat;

            for( i = 0; i < mPlaybackTracks.GetCount(); i++ )
            {
               // The mixer here isn't actually mixing: it's just doing
               // resampling, format conversion, and possibly time track
               // warping
               int processed = 0;
               samplePtr warpedSamples;
               //don't do anything if we have no length.  In particular, Process() will fail an wxAssert
               //that causes a crash since this is not the GUI thread and wxASSERT is a GUI call.
               if(deltat > 0.0)
               { 
                  processed = mPlaybackMixers[i]->Process(lrint(deltat * mRate));
                  warpedSamples = mPlaybackMixers[i]->GetBuffer();
                  mPlaybackBuffers[i]->Put(warpedSamples, floatSample, processed);
               }
               //if looping and processed is less than the full chunk/block/buffer that gets pulled from
               //other longer tracks, then we still need to advance the ring buffers or
               //we'll trip up on ourselves when we start them back up again.
               //if not looping we never start them up again, so its okay to not do anything
               if(processed < lrint(deltat * mRate) && mPlayLooped)
               {
                  if(mLastSilentBufSize < lrint(deltat * mRate))
                  {
                     //delete old if necessary
                     if(mSilentBuf) 
                        DeleteSamples(mSilentBuf);
                     mLastSilentBufSize=lrint(deltat * mRate);
                     mSilentBuf = NewSamples(mLastSilentBufSize, floatSample);
                     ClearSamples(mSilentBuf, floatSample, 0, mLastSilentBufSize);
                  }
                  mPlaybackBuffers[i]->Put(mSilentBuf, floatSample, lrint(deltat * mRate) - processed);                  
               }
            }

            // msmeyer: If playing looped, check if we are at the end of the buffer
            // and if yes, restart from the beginning.
            if (mPlayLooped && mT >= mWarpedT1)
            {
               for (i = 0; i < mPlaybackTracks.GetCount(); i++)
                  mPlaybackMixers[i]->Restart();
               mT = mT0;
            }

         } while (mPlayLooped && secsAvail > 0 && deltat > 0);
      }
   }  // end of playback buffering

   if( mCaptureTracks.GetCount() > 0 ) // start record buffering
   {
      int commonlyAvail = GetCommonlyAvailCapture();

      //
      // Determine how much this will add to captured tracks
      //
      double deltat = commonlyAvail / mRate;

      if (mAudioThreadShouldCallFillBuffersOnce ||
          deltat >= mMinCaptureSecsToCopy)
      {
         // Append captured samples to the end of the WaveTracks.
         // The WaveTracks have their own buffering for efficiency.
         XMLStringWriter blockFileLog;
         int numChannels = mCaptureTracks.GetCount();
         
         for( i = 0; (int)i < numChannels; i++ )
         {
            int avail = commonlyAvail;
            sampleFormat trackFormat = mCaptureTracks[i]->GetSampleFormat();

            XMLStringWriter appendLog;

            if( mFactor == 1.0 )
            {
               samplePtr temp = NewSamples(avail, trackFormat);
               mCaptureBuffers[i]->Get   (temp, trackFormat, avail);
               mCaptureTracks[i]-> Append(temp, trackFormat, avail, 1, 
                                          &appendLog);
               DeleteSamples(temp);
            }
            else
            {
               int size = lrint(avail * mFactor);
               samplePtr temp1 = NewSamples(avail, floatSample);
               samplePtr temp2 = NewSamples(size, floatSample);
               mCaptureBuffers[i]->Get(temp1, floatSample, avail);
               /* we are re-sampling on the fly. The last resampling call
                * must flush any samples left in the rate conversion buffer
                * so that they get recorded
                */
               size = mResample[i]->Process(mFactor, (float *)temp1, avail, !IsStreamActive(),
                                            &size, (float *)temp2, size);
               mCaptureTracks[i]-> Append(temp2, floatSample, size, 1, 
                                          &appendLog);
               DeleteSamples(temp1);
               DeleteSamples(temp2);
            }

            if (!appendLog.IsEmpty())
            {
               blockFileLog.StartTag(wxT("recordingrecovery"));
               blockFileLog.WriteAttr(wxT("channel"), (int)i);
               blockFileLog.WriteAttr(wxT("numchannels"), numChannels);
               blockFileLog.WriteSubTree(appendLog);
               blockFileLog.EndTag(wxT("recordingrecovery"));
            }
         }
         
         if (mListener && !blockFileLog.IsEmpty())
            mListener->OnAudioIONewBlockFiles(blockFileLog);
      }
   }  // end of record buffering

   //if ( mMidiStreamActive && mMidiPlaybackTracks.GetCount() > 0 )
      //FillMidiBuffers();
}

void AudioIO::SetListener(AudioIOListener* listener)
{
   if (IsBusy())
      return;

   mListener = listener;
}

#ifdef EXPERIMENTAL_MIDI_OUT

void AudioIO::OutputEvent()
{
   int channel = (mNextEvent->chan) & 0xF; // must be in [0..15]
   int command = -1;
   int data1 = -1;
   int data2 = -1;

   //   printf("OutputEvent: now %d\n", TIME_PROC(NULL));
         
   if (mVC & (1 << channel)) { // if mNextEvent's channel is visible, play it
      // Note event
      if (mNextEvent->is_note() && !mSendMidiState) {
         // Pitch and velocity
         data1 = mNextEvent->get_pitch();
         if (mNextIsNoteOn) data2 = mNextEvent->get_loud(); // get velocity
         else data2 = 0; // 0 velocity means "note off"
         command = 0x90; // MIDI NOTE ON (or OFF when velocity == 0)
      // Update event
      } else if (mNextEvent->is_update()) {
         // this code is based on allegrosmfwr.cpp -- it could be improved
         // by comparing attribute pointers instead of string compares
         Alg_update_ptr update = (Alg_update_ptr) mNextEvent;
         const char *name = update->get_attribute();
               
         if (!strcmp(name, "programi")) {
            // Instrument change
            data1 = update->parameter.i;
            data2 = 0;
            command = 0xC0; // MIDI PROGRAM CHANGE
         } else if (!strncmp(name, "control", 7)) {
            // Controller change

            // The number of the controller being changed is embedded
            // in the parameter name.
            data1 = atoi(name + 7);
            // Allegro normalizes controller values
            data2 = ROUND(update->parameter.r * 127);
            command = 0xB0;
         } else if (!strcmp(name, "bendr")) {
            // Bend change

            // Reverse Allegro's post-processing of bend values
            int temp = ROUND(0x2000 * (update->parameter.r + 1));
            if (temp > 0x3fff) temp = 0x3fff; // 14 bits maximum
            if (temp < 0) temp = 0;
            data1 = temp & 0x7f; // low 7 bits
            data2 = temp >> 7;   // high 7 bits
            command = 0xE0; // MIDI PITCH BEND
         } else if (!strcmp(name, "pressurer")) {
            // Pressure change
            data1 = (int) (update->parameter.r * 127);
            if (update->get_identifier() < 0) {
               // Channel pressure
               data2 = 0;
               command = 0xD0; // MIDI CHANNEL PRESSURE
            } else {
               // Key pressure
               data2 = data1;
               data1 = update->get_identifier();
               command = 0xA0; // MIDI POLY PRESSURE
            }
         }
      }
      if (command != -1) {
         // MIDI time is relative to the start of playback, but note times
         // are relative to the start of the track. mT0 is the correction
         // factor, because it's the time offset where playback started.
         double time = mNextEventTime - mT0;
         if (time < 0) time = 0;
         PmTimestamp timestamp = (PmTimestamp) (time * 1000); /* s to ms */
         Pm_WriteShort(mMidiStream, timestamp, 
                       Pm_Message((int) (command + channel), 
                                  (long) data1, (long) data2));
         // printf("midi out: time %d msg %2x %3d %3d\n",
         //       (PmTimestamp) (mNextEventTime * 1000), 
         //       (command + channel), (long) data1, (long) data2);
      }
   }
}

void AudioIO::GetNextEvent()
{
   mNextEvent = mIterator->next(&mNextIsNoteOn);
   if (mNextEvent) {
      if (mNextIsNoteOn) {
         mNextEventTime = mNextEvent->time;
      } else {
         mNextEventTime = mNextEvent->get_end_time();
      }
   } else {
      mLastMidiTime = (int) (mNextEventTime * 1000);
   }
   // printf("after GetNextEvent: time %g, event %p, mLastMidiTime %d\n", 
   //       mNextEventTime, mNextEvent, mLastMidiTime);
}

void AudioIO::FillMidiBuffers()
{
//   if (mPlaybackTracks.IsEmpty() && !mSendMidiState) {
//      gAudioIO->mTime = (now - mCurrentMidiTime)/1000 + mT0;
//   }

   // assume gAudioIO->mTime is the current time
   // assume output MIDI data 1s in advance of real time (mTime)
   // assume (for now) that midi time and sequence time are based at zero
   // then we need to iterate and output Midi until next time > mTime + 1s

   while (mNextEvent && 
          mNextEvent->time < gAudioIO->mTime + MIDI_BUFFER_AHEAD) {
      OutputEvent();
      GetNextEvent();
   }
#ifdef OLD_CODE
   if (now < mMidiWait) {
      return;
   }
   int i, j, k, track, visibleChannels;
   long channel, key, time;
   float command, data1, data2;
   double r;
   char updateParameter[13];
   bool forcedBreak = false;

   visibleChannels = mVC;

   if (mSeq) {
      i = 0; // index of buffer
      //testSeq->iteration_begin();

      Alg_event_ptr currEvent;

      while ( currEvent = mSeq->iteration_next() ) {
         // TODO HCK : this loop has Russian painter problem
         /*
         // In Update mode, events should be delivered immediately
         if (mSendMidiState) time = 0;
         // Normal playback mode takes the given event times
         else
            //time = (currEvent->time - mT0) * 1000;
            time = mCurrentMidiTime + ( currEvent->time - mT0 ) * 1000;
         */

         if (gAudioIO->mTime >= gAudioIO->mT1 && !gAudioIO->mPlayLooped) {
            mMidiStreamActive = false;
            // gAudioIO->mInCallbackFinishedState = true;
            mStreamToken = 0;
         }
         
         if (currEvent->time < mLastMidiTime / 1000) {
            continue;
         }

         if (currEvent->time >= mT0) {
            time = ( currEvent->time - mT0 ) * 1000 + mCurrentMidiTime;
            if (mSendMidiState) {
               mSendMidiState = false;
               mCurrentMidiTime = TIME_PROC(NULL);
            }
         } else {
            time = 0;
         }

         if (mCnt > 0) {
            j = 0;
            printf( "HCK : sorting...\n" );
            qsort( mMidiQueue, mCnt, sizeof( PmEvent ), compareTime );
            printf( "HCK : sorting...\n" );
            for (int azaa = 0; azaa < mCnt; azaa++) {
               printf( "HCK : SORT : %f %f\n", (float)mMidiQueue[azaa].timestamp, (float)time );
            }
            while (mMidiQueue[j].timestamp <= time) {
               mMidiBuffer[i].timestamp = mMidiQueue[j].timestamp;
               memcpy(&mMidiBuffer[i].message, &mMidiQueue[j].message,
                      sizeof(PmMessage));
               printf( "HCK : QUEUE!!!! : mCnt       : %d\n", mCnt );
               printf( "HCK : QUEUE!!!! : j          : %d\n", j );
               printf( "HCK : QUEUE!!!! : timestampQ : %f\n", (float)mMidiQueue[j].timestamp );
               printf( "HCK : QUEUE!!!! : timestampB : %f\n", (float)mMidiBuffer[i].timestamp );
               printf( "HCK : QUEUE!!!! : data Q     : %f\n", (float)mMidiQueue[j].message );
               printf( "HCK : QUEUE!!!! : data B     : %f\n", (float)mMidiBuffer[i].message );
               
               i++;
               j++;
               mCnt--;

               if (mCnt == 0) {
                  break;
               }
            }

            if (j > 0 && mCnt > 0) {
               memmove(&mMidiQueue[0], &mMidiQueue[j],
                       sizeof( PmEvent ) * mCnt );
            }
         }

         channel = currEvent->chan;
         command = data1 = data2 = -1;
         
         if (visibleChannels & (1 << channel)) {
            // Note event
            if (currEvent->is_note() && mSendMidiState == false) {
               // Pitch and velocity
               data1 = currEvent->get_pitch();
               data2 = currEvent->get_loud();
               command = 0x90;
            }
            // Update event
            else if (currEvent->is_update()) {
               // Allegro update events are stored as name/value parameters
               // where names can also contain important MIDI values and the
               // value data type.  To make this as easy as possible, we
               // only look at the first four characters of each name to 
               // determine the command.
               strcpy(updateParameter, 
                      ((Alg_update_ptr)currEvent)->parameter.attr_name());
               updateParameter[4] = 0;
               
               if (strcmp(updateParameter, "prog") == 0) {
                  // Instrument change
                  
                  data1 = ((Alg_update_ptr)currEvent)->parameter.i;
                  data2 = 0;
                  command = 0xC0;
               } else if (strcmp(updateParameter, "cont") == 0 &&
                          mSendMidiState == false) {
                  // Controller change

                  // The number of the controller being changed is embedded
                  // in the parameter name so we grab the whole name, set the
                  // index value to the position just after "control"
                  strcpy(updateParameter, 
                         ((Alg_update_ptr)currEvent)->parameter.attr_name());

                  k = 7;
                  data1 = 0;

                  while (updateParameter[k] != wxT('r')) {
                     data1 = data1 * 10 + atoi(&updateParameter[k]);
                     k++;
                  }

                  // Allegro normalizes controller values
                  data2 = ((Alg_update_ptr)currEvent)->parameter.r * 127;
                  command = 0xB0;
               } else if (strcmp(updateParameter, "bend") == 0 &&
                          mSendMidiState == false) {
                  // Bend change

                  // Reverse Allegro's post-processing of bend values
                  r = (((Alg_update_ptr)currEvent)->parameter.r + 1) * 8192;

                  data1 = ((long)r) >> 7;
                  data2 = (((long)r) << 7) >> 7;
                  command = 0xE0;
               } else if (strcmp(updateParameter, "pres") == 0 && 
                          mSendMidiState == false) {
                  // Pressure change

                  // Allegro normalizes pressures
                  r = ((Alg_update_ptr)currEvent)->parameter.r * 127;
                  key = currEvent->get_identifier();

                  // Channel pressure
                  if (key == -1) {
                     data1 = r;
                     data2 = 0;
                     command = 0xD0;
                  } else {
                     // Key pressure
                     data1 = key;
                     data2 = r;
                     command = 0xA0;
                  }
               }
            }
         }

         if (command != -1) {
            mMidiBuffer[i].timestamp = time;
            mMidiBuffer[i].message = Pm_Message((int)(command + channel), 
                                                (long)data1, (long)data2);
            printf( "HCK[%d]\n", i );
            printf( "Command     : %d\n", (int)command );
            printf( "mTime       : %f\n", (float)gAudioIO->mTime );
            printf( "TimeStamp   : %f\n", (float)mMidiBuffer[i].timestamp );
            printf( "CurMidiTime : %f\n", (float)mCurrentMidiTime );
            printf( "LastMidiTime: %f\n", (float)mLastMidiTime );
            printf( "MidiWait    : %f\n", (float)mMidiWait );
            printf( "Time        : %f\n", (float)TIME_PROC(NULL) );
            i++;
            if (command == 0x90) {
               mMidiQueue[mCnt].timestamp =
                  time + (long)currEvent->get_duration() * 1000;
               mMidiQueue[mCnt].message = Pm_Message((int)(0x90 + channel),
                                                     (long)data1, 0 );
               printf( "HCK QUEUE[%d]\n", mCnt );
               printf( "Command     : OFF\n" );
               printf( "mTime       : %f\n", (float)gAudioIO->mTime );
               printf( "TimeStamp   : %f\n", (float)mMidiQueue[mCnt].timestamp );
               printf( "CurMidiTime : %f\n", (float)mCurrentMidiTime );
               printf( "LastMidiTime: %f\n", (float)mLastMidiTime );
               printf( "MidiWait    : %f\n", (float)mMidiWait );
               printf( "Time        : %f\n", (float)TIME_PROC(NULL) );
               mCnt++;
            } else
               fprintf(stderr, "command: %s\n", updateParameter);
         }
         
         // Turn off updates when we reach the selection beginning
         /*
           if (mSendMidiState) {
              if (i == 0 && i > testSeq->seek_time(mT0, track)) {
                 // The first 1/10 of the file has been processed
                 // so just to 5 seconds before cursor to avoid lag
                 notesOn = true;
                 i = testSeq->seek_time(mT0, track);
                 fprintf(stderr, "%li: Stop processing updates\n", TIME_PROC(NULL));
              }
           }
         */
         // Stop when:
         // 1. enough events are buffered
         // 2. there are no more events to buffer
         //if (i >= endIndex || i == testSeq->length())
         if (i >= MAX_MIDI_BUFFER_SIZE - 1) {
            if (!mSendMidiState) {
               printf( "HCK : Pm_Write : 111111111\n" );
               qsort( mMidiBuffer, i, sizeof( PmEvent ), compareTime );
               Pm_Write(mMidiStream, mMidiBuffer, i);
               mMidiWait = time - 1000;
            }
            i = 0;
            mLastMidiTime = currEvent->time * 1000;
            forcedBreak = true;
            break;
         } else if (currEvent->time * 1000 >= mLastMidiTime + 2000) {
            if (!mSendMidiState) {
               printf( "HCK : Pm_Write : 222222222\n" );
               qsort( mMidiBuffer, i, sizeof( PmEvent ), compareTime );
               Pm_Write( mMidiStream, mMidiBuffer, i );
               mMidiWait = time - 1000;
            }
            i = 0;
            mLastMidiTime = currEvent->time * 1000;
            forcedBreak = true;
            break;
         }
      }  // End of While
         //testSeq->iteration_end();
      if (!forcedBreak ) { // this means there are no more event in testSeq.
         mMidiStreamActive = false;
         // gAudioIO->mInCallbackFinishedState = true;
      }
   } // End of if( mSeq )
#endif // OLD_CODE
}

void AudioIO::AudacityMidiCallback() {

   // instrumentation to see if we are called often

   // This calculation is taken from GetStreamTime, found in pa_asio.cpp
   // asio systemTime is supposed to be measured according to the same
   // clock as timeGetTime
   double currentSystemTime = (double)timeGetTime() * .001; // get current system time

   if(!mLastSystemTime) mLastSystemTime = currentSystemTime;
   if(!mLatencyBetweenSystemTimes) mLatencyBetweenSystemTimes = 0;

   mLatencyBetweenSystemTimes = max(mLatencyBetweenSystemTimes, currentSystemTime - mLastSystemTime);
   mLastMidiTime = currentSystemTime;

   if(mRequestMidiStop) {
      /* TODO: Send all notes off */
      mMidiStreamActive = false;
      return;
   }
   if(mPaused || !mMidiStreamActive) { return; }

   //look at next message
   //assume next message from iterator is available
   //(so create iterator & get/store first event before
   //callback is called)

   double t = getCurrentTrackTime();
   while(mNextEvent != NULL && mNextEvent->time < t + 0.2) {
      long ts = calculateMidiTimeStamp(mNextEvent->time);

      /* TODO: Where is ts supposed to be used?
       * send next message to portmidi with ts;
       * mNextEvent = mIterator->next();
       */
      OutputEvent();
      GetNextEvent();
   }
}

double AudioIO::getCurrentTrackTime() {
   //compare to code that scrolls cursor

   // This calculation is taken from GetStreamTime, found in pa_asio.cpp
   // asio systemTime is supposed to be measured according to the same
   // clock as timeGetTime
   double timeSinceBoot = (double)timeGetTime() * .001; // get time since boot

   double outputTime;
   long sampleNumber;
   do {
      outputTime = mAudioCallbackOutputTime;
      sampleNumber = mAudioCallbackSampleNumber;
   } while (outputTime != mAudioCallbackOutputTime);
   double delta = timeSinceBoot - outputTime;

   mStartFrame = mT0 * mRate;

   // We add startFrame in case we're not starting at the beginning of the track
   double trackTime = (sampleNumber + mStartFrame - mNumPauseFrames) / mRate;

   // at outputTime, trackTime will be correct

   trackTime += delta;
   //now tracktime is adjusted from acot to now
   return trackTime;
}

long AudioIO::calculateMidiTimeStamp(double eventTime) {
   //assumptions: portmidi uses default time function which is "system time" - start time
   double outputTime;
   long sampleNumber;
   do {
      outputTime = mAudioCallbackOutputTime;
      sampleNumber = mAudioCallbackSampleNumber;
   } while (outputTime != mAudioCallbackOutputTime);

   mStartFrame = mT0 * mRate;

   double trackTime = mAudioCallbackSampleNumber + mStartFrame - mNumPauseFrames / mRate;
   //at ot, tracktime will be correct
   double delta = eventTime - trackTime;

   outputTime += delta;  // at outputTime, it will be time for event
   outputTime -= Pt_Time(); // get startTime from PortTime library

   // now, outputTime is in PortMidi timestamp coordinates

   long outputTimeMS = (outputTime * 1000) - 100; // convert to ms and subtract PortMidi latency
   return outputTimeMS;
}


#endif

// Automated Input Level Adjustment - Automatically tries to find an acceptable input volume
#ifdef AUTOMATED_INPUT_LEVEL_ADJUSTMENT
void AudioIO::AILAInitialize() {
   gPrefs->Read(wxT("/AudioIO/AutomatedInputLevelAdjustment"), &mAILAActive,         false);
   gPrefs->Read(wxT("/AudioIO/TargetPeak"),            &mAILAGoalPoint,      AILA_DEF_TARGET_PEAK);
   gPrefs->Read(wxT("/AudioIO/DeltaPeakVolume"),       &mAILAGoalDelta,      AILA_DEF_DELTA_PEAK);
   gPrefs->Read(wxT("/AudioIO/AnalysisTime"),          &mAILAAnalysisTime,   AILA_DEF_ANALYSIS_TIME);
   gPrefs->Read(wxT("/AudioIO/NumberAnalysis"),        &mAILATotalAnalysis,  AILA_DEF_NUMBER_ANALYSIS);
   mAILAGoalDelta         /= 100.0;
   mAILAGoalPoint         /= 100.0; 
   mAILAAnalysisTime      /= 1000.0;
   mAILAMax                = 0.0;
   mAILALastStartTime      = max(0.0, mT0);
   mAILAClipped            = false;
   mAILAAnalysisCounter    = 0;
   mAILAChangeFactor       = 1.0;
   mAILALastChangeType     = 0;
   mAILATopLevel           = 1.0;
   mAILAAnalysisEndTime    = -1.0;
}

void AudioIO::AILADisable() {
   mAILAActive = false;
}

bool AudioIO::AILAIsActive() {
   return mAILAActive;
}

void AudioIO::AILASetStartTime() {
   mAILAAbsolutStartTime = Pa_GetStreamTime(mPortStreamV19);
   printf("START TIME %f\n\n", mAILAAbsolutStartTime);
}

double AudioIO::AILAGetLastDecisionTime() {
   return mAILAAnalysisEndTime;
}

void AudioIO::AILAProcess(double maxPeak) {
   AudacityProject *proj = GetActiveProject();
   if (proj && mAILAActive) {
      if (mInputMeter->IsClipping()) {
         mAILAClipped = true;
         printf("clipped");
      }
      
      mAILAMax = max(mAILAMax, maxPeak);
   
      if ((mAILATotalAnalysis == 0 || mAILAAnalysisCounter < mAILATotalAnalysis) && mTime - mAILALastStartTime >= mAILAAnalysisTime) {
         putchar('\n');
         mAILAMax = mInputMeter->ToLinearIfDB(mAILAMax);
         double iv = (double) Px_GetInputVolume(mPortMixer);
         unsigned short changetype = 0; //0 - no change, 1 - increase change, 2 - decrease change
         printf("mAILAAnalysisCounter:%d\n", mAILAAnalysisCounter);
         printf("\tmAILAClipped:%d\n", mAILAClipped);
         printf("\tmAILAMax (linear):%f\n", mAILAMax);
         printf("\tmAILAGoalPoint:%f\n", mAILAGoalPoint);
         printf("\tmAILAGoalDelta:%f\n", mAILAGoalDelta);
         printf("\tiv:%f\n", iv);
         printf("\tmAILAChangeFactor:%f\n", mAILAChangeFactor);
         if (mAILAClipped || mAILAMax > mAILAGoalPoint + mAILAGoalDelta) {
            printf("too high:\n");
            mAILATopLevel = min(mAILATopLevel, iv);
            printf("\tmAILATopLevel:%f\n", mAILATopLevel);
            //if clipped or too high
            if (iv <= LOWER_BOUND) {
               //we can't improve it more now
               if (mAILATotalAnalysis != 0) {
                  mAILAActive = false;
                  proj->TP_DisplayStatusMessage(_("Automated Input Level Adjustment stopped. It was not possible to optimize it more. Still too high."));
               }
               printf("\talready min vol:%f\n", iv);
            }
            else {
               float vol = (float) max(LOWER_BOUND, iv+(mAILAGoalPoint-mAILAMax)*mAILAChangeFactor);
               Px_SetInputVolume(mPortMixer, vol);
               wxString msg;
               msg.Printf(_("Automated Input Level Adjustment decreased the volume to %f."), vol);
               proj->TP_DisplayStatusMessage(msg);
               changetype = 1;
               printf("\tnew vol:%f\n", vol);
               float check = Px_GetInputVolume(mPortMixer);
               printf("\tverified %f\n", check);
            }
         }
         else if ( mAILAMax < mAILAGoalPoint - mAILAGoalDelta ) {
            //if too low
            printf("too low:\n"); 
            if (iv >= UPPER_BOUND || iv + 0.005 > mAILATopLevel) { //condition for too low volumes and/or variable volumes that cause mAILATopLevel to decrease too much
               //we can't improve it more
               if (mAILATotalAnalysis != 0) {
                  mAILAActive = false;
                  proj->TP_DisplayStatusMessage(_("Automated Input Level Adjustment stopped. It was not possible to optimize it more. Still too low."));
               }
               printf("\talready max vol:%f\n", iv);
            }
            else {
               float vol = (float) min(UPPER_BOUND, iv+(mAILAGoalPoint-mAILAMax)*mAILAChangeFactor);
               if (vol > mAILATopLevel) {
                  vol = (iv + mAILATopLevel)/2.0;
                  printf("\tTruncated vol:%f\n", vol);
               }
               Px_SetInputVolume(mPortMixer, vol);
               wxString msg;
               msg.Printf(_("Automated Input Level Adjustment increased the volume to %.2f."), vol);
               proj->TP_DisplayStatusMessage(msg);
               changetype = 2;
               printf("\tnew vol:%f\n", vol);
               float check = Px_GetInputVolume(mPortMixer);
               printf("\tverified %f\n", check);
            }
         }

         mAILAAnalysisCounter++;
         //const PaStreamInfo* info = Pa_GetStreamInfo(mPortStreamV19);
         //double latency = 0.0;
         //if (info)
         //   latency = info->inputLatency;
         //mAILAAnalysisEndTime = mTime+latency;
         mAILAAnalysisEndTime = Pa_GetStreamTime(mPortStreamV19) - mAILAAbsolutStartTime;
         mAILAMax             = 0;
         printf("\tA decision was made @ %f\n", mAILAAnalysisEndTime);
         mAILAClipped         = false;  
         mAILALastStartTime   = mTime;
         
         if (changetype == 0)
            mAILAChangeFactor *= 0.8; //time factor
         else if (mAILALastChangeType == changetype)
            mAILAChangeFactor *= 1.1; //concordance factor
         else
            mAILAChangeFactor *= 0.7; //discordance factor
         mAILALastChangeType = changetype;
         putchar('\n');
      }

      if (mAILAActive && mAILATotalAnalysis != 0 && mAILAAnalysisCounter >= mAILATotalAnalysis) {
         mAILAActive = false;
         if (mAILAMax > mAILAGoalPoint + mAILAGoalDelta)
            proj->TP_DisplayStatusMessage(_("Automated Input Level Adjustment stopped. The total number of analysis has been exceeded without finding an acceptable volume. Still too high."));
         else if (mAILAMax < mAILAGoalPoint - mAILAGoalDelta)
            proj->TP_DisplayStatusMessage(_("Automated Input Level Adjustment stopped. The total number of analysis has been exceeded without finding an acceptable volume. Still too low."));
         else {
            wxString msg;
            msg.Printf(_("Automated Input Level Adjustment stopped. %.2f seems an acceptable volume."), Px_GetInputVolume(mPortMixer));
            proj->TP_DisplayStatusMessage(msg);
         }
      }
   }
}
#endif

//////////////////////////////////////////////////////////////////////
//
//    PortAudio callback thread context
//
//////////////////////////////////////////////////////////////////////

#define MAX(a,b) ((a) > (b) ? (a) : (b))

void DoSoftwarePlaythrough(const void *inputBuffer,
                           sampleFormat inputFormat,
                           int inputChannels,
                           float *outputBuffer,
                           int len,
                           float gain)
{
   float *tempBuffer = (float *)alloca(len * sizeof(float));
   int i, j;

   for(j=0; j<inputChannels; j++) {
      samplePtr inputPtr = ((samplePtr)inputBuffer) + (j * SAMPLE_SIZE(inputFormat));
      
      CopySamples(inputPtr, inputFormat,
                  (samplePtr)tempBuffer, floatSample,
                  len, true, inputChannels);

      for(i=0; i<len; i++)
         outputBuffer[2*i + (j%2)] = tempBuffer[i];

      // One mono input channel goes to both output channels...
      if (inputChannels == 1)
         for(i=0; i<len; i++)
            outputBuffer[2*i + 1] = tempBuffer[i];
   }
   
}

int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
                          unsigned long framesPerBuffer,
                          const PaStreamCallbackTimeInfo *timeInfo,
                          const PaStreamCallbackFlags statusFlags, void *userData )
{
   int numPlaybackChannels = gAudioIO->mNumPlaybackChannels;
   int numPlaybackTracks = gAudioIO->mPlaybackTracks.GetCount();
   int numCaptureChannels = gAudioIO->mNumCaptureChannels;
   int callbackReturn = paContinue;
   void *tempBuffer = alloca(framesPerBuffer*sizeof(float)*
                             MAX(numCaptureChannels,numPlaybackChannels));
   float *tempFloats = (float*)tempBuffer;

   // output meter may need samples untouched by volume emulation
   float *outputMeterFloats;
   outputMeterFloats =
      (outputBuffer && gAudioIO->mEmulateMixerOutputVol &&
                       gAudioIO->mMixerOutputVol != 1.0) ?
         (float *)alloca(framesPerBuffer*numPlaybackChannels * sizeof(float)) :
         (float *)outputBuffer;

#ifdef EXPERIMENTAL_MIDI_OUT
   /* GSW: Save timeInfo in case MidiPlayback needs it */
   gAudioIO->mAudioCallbackOutputTime = timeInfo->outputBufferDacTime;
   gAudioIO->mAudioCallbackSampleNumber += framesPerBuffer;
   if(gAudioIO->IsPaused())
      gAudioIO->mNumPauseFrames += framesPerBuffer;
#endif

   unsigned int i;
   int t;

   /* Send data to recording VU meter if applicable */

   if (gAudioIO->mInputMeter &&
         !gAudioIO->mInputMeter->IsMeterDisabled() &&
         inputBuffer) {
      // get here if meters are actually live , and being updated
      /* It's critical that we don't update the meters while StopStream is
       * trying to stop PortAudio, otherwise it can lead to a freeze.  We use
       * two variables to synchronize:
       *   mUpdatingMeters tells StopStream when the callback is about to enter
       *     the code where it might update the meters, and
       *   mUpdateMeters is how the rest of the code tells the callback when it
       *     is allowed to actually do the updating.
       * Note that mUpdatingMeters must be set first to avoid a race condition.
       */
      gAudioIO->mUpdatingMeters = true;
      if (gAudioIO->mUpdateMeters) {
         if (gAudioIO->mCaptureFormat == floatSample)
            gAudioIO->mInputMeter->UpdateDisplay(numCaptureChannels,
                                                 framesPerBuffer,
                                                 (float *)inputBuffer);
         else {
            CopySamples((samplePtr)inputBuffer, gAudioIO->mCaptureFormat,
                        (samplePtr)tempFloats, floatSample,
                        framesPerBuffer * numCaptureChannels);
            gAudioIO->mInputMeter->UpdateDisplay(numCaptureChannels,
                                                 framesPerBuffer,
                                                 tempFloats);
         }
      }
      gAudioIO->mUpdatingMeters = false;
   }  // end recording VU meter update

   // Stop recording if 'silence' is detected
   if(gAudioIO->mPauseRec && inputBuffer && gAudioIO->mInputMeter) {
      if(gAudioIO->mInputMeter->GetMaxPeak() < gAudioIO->mSilenceLevel ) {
         if(!gAudioIO->IsPaused()) {
            AudacityProject *p = GetActiveProject();
            wxCommandEvent dummyEvt;
            p->GetControlToolBar()->OnPause(dummyEvt);
         }
      }
      else {
         if(gAudioIO->IsPaused()) {
            AudacityProject *p = GetActiveProject();
            wxCommandEvent dummyEvt;
            p->GetControlToolBar()->OnPause(dummyEvt);
         }
      }
   }
   if( gAudioIO->mPaused )
   {
      if (outputBuffer && numPlaybackChannels > 0)
      {
         ClearSamples((samplePtr)outputBuffer, floatSample,
                      0, framesPerBuffer * numPlaybackChannels);

         if (inputBuffer && gAudioIO->mSoftwarePlaythrough) {
            DoSoftwarePlaythrough(inputBuffer, gAudioIO->mCaptureFormat,
                                  numCaptureChannels,
                                  (float *)outputBuffer, (int)framesPerBuffer, 1.0f);
         }
      }

      return paContinue;
   }

   if (gAudioIO->mStreamToken > 0)
   {
      //
      // Mix and copy to PortAudio's output buffer
      //
      
      if( outputBuffer && (numPlaybackChannels > 0) )
      {
         bool cut = false;
         bool linkFlag = false;
         
         float *outputFloats = (float *)outputBuffer;
         for( i = 0; i < framesPerBuffer*numPlaybackChannels; i++)
            outputFloats[i] = 0.0;

         if (inputBuffer && gAudioIO->mSoftwarePlaythrough) {
            DoSoftwarePlaythrough(inputBuffer, gAudioIO->mCaptureFormat,
                                  numCaptureChannels,
                                  (float *)outputBuffer, (int)framesPerBuffer, 1.0f);
         }

         // Copy the results to outputMeterFloats if necessary
         if (outputMeterFloats != outputFloats) {
            for (i = 0; i < framesPerBuffer*numPlaybackChannels; ++i) {
               outputMeterFloats[i] = outputFloats[i];
            }
         }

         if (gAudioIO->mSeek)
         {
            // Pause audio thread and wait for it to finish
            gAudioIO->mAudioThreadFillBuffersLoopRunning = false;
            while( gAudioIO->mAudioThreadFillBuffersLoopActive == true )
            {
               wxMilliSleep( 50 );
            }

            // Calculate the new time position
            gAudioIO->mTime += gAudioIO->mSeek;
            if (gAudioIO->mTime < gAudioIO->mT0)
                gAudioIO->mTime = gAudioIO->mT0;
            else if (gAudioIO->mTime > gAudioIO->mT1)
                gAudioIO->mTime = gAudioIO->mT1;
            gAudioIO->mSeek = 0.0;
            
            // Reset mixer positions and flush buffers for all tracks
            gAudioIO->mT = gAudioIO->mT0 + ((gAudioIO->mTime - gAudioIO->mT0));
            for (i = 0; i < (unsigned int)numPlaybackTracks; i++)
            {
               gAudioIO->mPlaybackMixers[i]->Reposition(gAudioIO->mT);
               gAudioIO->mPlaybackBuffers[i]->Discard(gAudioIO->mPlaybackBuffers[i]->AvailForGet());
            }

            // Reload the ring buffers
            gAudioIO->mAudioThreadShouldCallFillBuffersOnce = true;
            while( gAudioIO->mAudioThreadShouldCallFillBuffersOnce == true )
            {
               wxMilliSleep( 50 );
            }

            // Reenable the audio thread
            gAudioIO->mAudioThreadFillBuffersLoopRunning = true;
               
            return paContinue;
         }

         int numSolo = 0;
         for( t = 0; t < numPlaybackTracks; t++ )
            if( gAudioIO->mPlaybackTracks[t]->GetSolo() )
               numSolo++;
         
         for( t = 0; t < numPlaybackTracks; t++)
         {
            WaveTrack *vt = gAudioIO->mPlaybackTracks[t];
            
            if (linkFlag)
               linkFlag = false;
            else {
               cut = false;
               
               // Cut if somebody else is soloing
               if (numSolo>0 && !vt->GetSolo())
                  cut = true;
               
               // Cut if we're muted (unless we're soloing)
               if (vt->GetMute() && !vt->GetSolo())
                  cut = true;
               
               linkFlag = vt->GetLinked();
            }
            
            if (cut)
               {
                  gAudioIO->mPlaybackBuffers[t]->Discard(framesPerBuffer);
                  continue;
               }

            unsigned int len = (unsigned int)
               gAudioIO->mPlaybackBuffers[t]->Get((samplePtr)tempFloats,
                                                  floatSample,
                                                  (int)framesPerBuffer);

            // If our buffer is empty and the time indicator is past
            // the end, then we've actually finished playing the entire
            // selection.
            // msmeyer: We never finish if we are playing looped
            if (len == 0 && gAudioIO->mTime >= gAudioIO->mT1 &&
                !gAudioIO->mPlayLooped)
            {
               callbackReturn = paComplete;
            }

            if (vt->GetChannel() == Track::LeftChannel ||
                vt->GetChannel() == Track::MonoChannel)
            {
               float gain = vt->GetChannelGain(0);

               // Output volume emulation: possibly copy meter samples, then
               // apply volume, then copy to the output buffer
               if (outputMeterFloats != outputFloats)
                  for (i = 0; i < len; ++i)
                     outputMeterFloats[numPlaybackChannels*i] +=
                        gain*tempFloats[i];
               
               if (gAudioIO->mEmulateMixerOutputVol)
                  gain *= gAudioIO->mMixerOutputVol;
               
               for(i=0; i<len; i++)
                  outputFloats[numPlaybackChannels*i] += gain*tempFloats[i];
            }
            
            if (vt->GetChannel() == Track::RightChannel ||
                vt->GetChannel() == Track::MonoChannel)
            {
               float gain = vt->GetChannelGain(1);
               
               // Output volume emulation (as above)
               if (outputMeterFloats != outputFloats)
                  for (i = 0; i < len; ++i)
                     outputMeterFloats[numPlaybackChannels*i+1] +=
                        gain*tempFloats[i];

               if (gAudioIO->mEmulateMixerOutputVol)
                  gain *= gAudioIO->mMixerOutputVol;
               
               for(i=0; i<len; i++)
                  outputFloats[numPlaybackChannels*i+1] += gain*tempFloats[i];
            }
         }
         
         //
         // Clip output to [-1.0,+1.0] range (msmeyer)
         //
         for( i = 0; i < framesPerBuffer*numPlaybackChannels; i++)
         {
            float f = outputFloats[i];
            if (f > 1.0)
               outputFloats[i] = 1.0;
            else if (f < -1.0)
               outputFloats[i] = -1.0;
         }

         // Same for meter output
         if (outputMeterFloats != outputFloats)
         {
            for (i = 0; i < framesPerBuffer*numPlaybackChannels; ++i)
            {
               float f = outputMeterFloats[i];
               if (f > 1.0)
                  outputMeterFloats[i] = 1.0;
               else if (f < -1.0)
                  outputMeterFloats[i] = -1.0;
            }
         }
      }

      //
      // Copy from PortAudio to our input buffers.
      //
      
      if( inputBuffer && (numCaptureChannels > 0) )
      {
         unsigned int len = framesPerBuffer;
         for( t = 0; t < numCaptureChannels; t++) {
            unsigned int avail =
               (unsigned int)gAudioIO->mCaptureBuffers[t]->AvailForPut();
            if (avail < len)
               len = avail;
         }
         
         if (len < framesPerBuffer)
         {
            gAudioIO->mLostSamples += (framesPerBuffer - len);
            wxPrintf(wxT("lost %d samples\n"), (int)(framesPerBuffer - len));
         }

         if (len > 0) {
            for( t = 0; t < numCaptureChannels; t++) {
               
               // dmazzoni:
               // Un-interleave.  Ugly special-case code required because the
               // capture channels could be in three different sample formats;
               // it'd be nice to be able to call CopySamples, but it can't
               // handle multiplying by the gain and then clipping.  Bummer.

               switch(gAudioIO->mCaptureFormat) {
               case floatSample: {
                  float *inputFloats = (float *)inputBuffer;
                  for( i = 0; i < len; i++)
                     tempFloats[i] =
                        inputFloats[numCaptureChannels*i+t];
               } break;
               case int24Sample:
                  // We should never get here. Audacity's int24Sample format
                  // is different from PortAudio's sample format and so we
                  // make PortAudio return float samples when recording in
                  // 24-bit samples.
                  wxASSERT(false);
                  break;
               case int16Sample: {
                  short *inputShorts = (short *)inputBuffer;
                  short *tempShorts = (short *)tempBuffer;
                  for( i = 0; i < len; i++) {
                     float tmp = inputShorts[numCaptureChannels*i+t];
                     if (tmp > 32767)
                        tmp = 32767;
                     if (tmp < -32768)
                        tmp = -32768;
                     tempShorts[i] = (short)(tmp);
                  }
               } break;
               } // switch
               
               gAudioIO->mCaptureBuffers[t]->Put((samplePtr)tempBuffer,
                                                 gAudioIO->mCaptureFormat,
                                                 len);
            }
         }
      }

      // Calcuate the warp factor for this time position
      double factor = 1.0;
      if (gAudioIO->mTimeTrack) {
         factor = gAudioIO->mTimeTrack->GetEnvelope()->GetValue(gAudioIO->mTime);
         factor = (gAudioIO->mTimeTrack->GetRangeLower() *
                  (1 - factor) +
                  factor *
                  gAudioIO->mTimeTrack->GetRangeUpper()) / 
                  100.0;
      }

      // Wrap to start if looping
      if (gAudioIO->mPlayLooped && gAudioIO->mTime >= gAudioIO->mT1)
      {
         // LL:  This is not exactly right, but I'm at my wits end trying to
         //      figure it out.  Feel free to fix it.  :-)
         gAudioIO->mTime = gAudioIO->mT0 - ((gAudioIO->mTime - gAudioIO->mT1) * factor);
      }

      // Update the current time position
      gAudioIO->mTime += ((framesPerBuffer / gAudioIO->mRate) * factor);

      // Record the reported latency from PortAudio.
      // TODO: Don't recalculate this with every callback?

      // 01/21/2009:  Disabled until a better solution presents itself.
     #if 0
      // As of 06/17/2006, portaudio v19 returns inputBufferAdcTime set to
      // zero.  It is being worked on, but for now we just can't do much
      // but follow the leader.
      //
      // 08/27/2006: too inconsistent for now...just leave it a zero.
      //
      // 04/16/2008: Looks like si->inputLatency comes back with something useful though.
      // This rearranged logic uses si->inputLatency, but if PortAudio fixes inputBufferAdcTime, 
      // this code won't have to be modified to use it. 
      // Also avoids setting mLastRecordingOffset except when simultaneously playing and recording.
      //
      if (numCaptureChannels > 0 && numPlaybackChannels > 0) // simultaneously playing and recording
      {
         if (timeInfo->inputBufferAdcTime > 0)
            gAudioIO->mLastRecordingOffset = timeInfo->inputBufferAdcTime - timeInfo->outputBufferDacTime;
         else if (gAudioIO->mLastRecordingOffset == 0.0) 
         {
            const PaStreamInfo* si = Pa_GetStreamInfo( gAudioIO->mPortStreamV19 );
            gAudioIO->mLastRecordingOffset = -si->inputLatency;
         }
      }
     #endif
   } // if mStreamToken > 0
   else {
      // No tracks to play, but we should clear the output, and
      // possibly do software playthrough...
      
      if( outputBuffer && (numPlaybackChannels > 0) ) {
         float *outputFloats = (float *)outputBuffer;
         for( i = 0; i < framesPerBuffer*numPlaybackChannels; i++)
            outputFloats[i] = 0.0;
         
         if (inputBuffer && gAudioIO->mSoftwarePlaythrough) {
            DoSoftwarePlaythrough(inputBuffer, gAudioIO->mCaptureFormat,
                                  numCaptureChannels,
                                  (float *)outputBuffer, (int)framesPerBuffer, 1.0f);
         }

         // Copy the results to outputMeterFloats if necessary
         if (outputMeterFloats != outputFloats) {
            for (i = 0; i < framesPerBuffer*numPlaybackChannels; ++i) {
               outputMeterFloats[i] = outputFloats[i];
            }
         }
      }

   }
   /* Send data to playback VU meter if applicable */
   if (gAudioIO->mOutputMeter && 
      !gAudioIO->mOutputMeter->IsMeterDisabled() &&
      outputMeterFloats) {
      // Get here if playback meter is live 
      /* It's critical that we don't update the meters while StopStream is
       * trying to stop PortAudio, otherwise it can lead to a freeze.  We use
       * two variables to synchronize:
       *  mUpdatingMeters tells StopStream when the callback is about to enter
       *    the code where it might update the meters, and 
       *  mUpdateMeters is how the rest of the code tells the callback when it
       *    is allowed to actually do the updating.
       * Note that mUpdatingMeters must be set first to avoid a race condition.
       */
      gAudioIO->mUpdatingMeters = true;
      if (gAudioIO->mUpdateMeters) {
         gAudioIO->mOutputMeter->UpdateDisplay(numPlaybackChannels,
                                               framesPerBuffer,
                                               outputMeterFloats);
      }
      gAudioIO->mUpdatingMeters = false;
   }  // end playback VU meter update

   return callbackReturn;
}

#ifdef EXPERIMENTAL_MIDI_OUT
int compareTime( const void* a, const void* b )
{
   return( (int)((*(PmEvent*)a).timestamp - (*(PmEvent*)b).timestamp ) );
}
#endif

// Indentation settings for Vim and Emacs and unique identifier for Arch, a
// version control system. Please do not modify past this point.
//
// Local Variables:
// c-basic-offset: 3
// indent-tabs-mode: nil
// End:
//
// vim: et sts=3 sw=3
// arch-tag: 7ee3c9aa-b58b-4069-8a07-8866f2303963