C++ WaveTrack::GetStartTime方法代码示例

bool EffectChangeSpeed::Process()
{
	// Similar to EffectSoundTouch::Process()

   //Iterate over each track
   TrackListIterator iter(mWaveTracks);
   WaveTrack *track = (WaveTrack *) iter.First();
   mCurTrackNum = 0;
	m_maxNewLength = 0.0;
	double curT0;
	double curT1;
   while (track) {
      //Get start and end times from track
      double trackStart = track->GetStartTime();
      double trackEnd = track->GetEndTime();

      //Set the current bounds to whichever left marker is
      //greater and whichever right marker is less:
      curT0 = mT0 < trackStart? trackStart: mT0;
      curT1 = mT1 > trackEnd? trackEnd: mT1;

      // Process only if the right marker is to the right of the left marker
      if (curT1 > curT0) {

         //Transform the marker timepoints to samples
         longSampleCount start = track->TimeToLongSamples(curT0);
         longSampleCount end = track->TimeToLongSamples(curT1);

         //ProcessOne() (implemented below) processes a single track
         if (!ProcessOne(track, start, end))
            return false;
      }
      
      //Iterate to the next track
      track = (WaveTrack *) iter.Next();
      mCurTrackNum++;
   }

	mT1 = mT0 + m_maxNewLength; // Update selection.
   return true;
}

bool EffectNoiseRemoval::Process()
{
	// If we are creating a profile, we don't care whether we have
	// one already.  We just prepare the counters.
   if (mDoProfile) {
      for(int i=0; i<windowSize; i++) {
         sum[i] = float(0.0);
         sumsq[i] = float(0.0);
         profileCount[i] = 0;
      }
   }
	else
	{
		// We need a profile.
      if( !mHasProfile )
      {
         CleanSpeechMayReadNoisegate();
      }
   
	   // If we still don't have a profile we have a problem.
      if( !mHasProfile)
      {
         wxMessageBox( _("Attempt to run Noise Removal without a noise profile\n.") );
         return false;
      }
	}

	// This same code will both remove noise and
	// profile it, depending on 'mDoProfile'
   TrackListIterator iter(mWaveTracks);
   WaveTrack *track = (WaveTrack *) iter.First();
   int count = 0;
   while (track) {
      double trackStart = track->GetStartTime();
      double trackEnd = track->GetEndTime();
      double t0 = mT0 < trackStart? trackStart: mT0;
      double t1 = mT1 > trackEnd? trackEnd: mT1;

      if (t1 > t0) {
         longSampleCount start = track->TimeToLongSamples(t0);
         longSampleCount end = track->TimeToLongSamples(t1);
         sampleCount len = (sampleCount)(end - start);

         if (!ProcessOne(count, track, start, len)){
            return false;
	      }
      }
      track = (WaveTrack *) iter.Next();
      count++;
   }

   if (mDoProfile) {
      for(int i=0; i<=windowSize/2; i++) {
         //float stddev = sqrt(sumsq[i] - (sum[i]*sum[i])/profileCount[i])
         //                               / profileCount[i];
         mNoiseGate[i] = sum[i] / profileCount[i]; // average
      }
		CleanSpeechMayWriteNoiseGate();
      mHasProfile = true;
      mDoProfile = false;
   }
   return true;
}

void Effect::Preview()
{
   wxWindow* FocusDialog = wxWindow::FindFocus();
   if (gAudioIO->IsBusy())
      return;

   // Mix a few seconds of audio from all of the tracks
   double previewLen = 6.0;
   gPrefs->Read(wxT("/AudioIO/EffectsPreviewLen"), &previewLen);
   
   WaveTrack *mixLeft = NULL;
   WaveTrack *mixRight = NULL;
   double rate = mProjectRate;
   double t0 = mT0;
   double t1 = t0 + previewLen;

   if (t1 > mT1)
      t1 = mT1;

   if (t1 <= t0)
      return;

   bool success = ::MixAndRender(mTracks, mFactory, rate, floatSample, t0, t1,
                                 &mixLeft, &mixRight);

   if (!success) {
      return;
   }

   // Save the original track list
   TrackList *saveTracks = mTracks;

   // Build new tracklist from rendering tracks
   mTracks = new TrackList();
   mixLeft->SetSelected(true);   
   mTracks->Add(mixLeft);
   if (mixRight) {
      mixRight->SetSelected(true);   
      mTracks->Add(mixRight);
   }

   // Update track/group counts
   CountWaveTracks();

   // Reset times
   t0 = mixLeft->GetStartTime();
   t1 = mixLeft->GetEndTime();

   double t0save = mT0;
   double t1save = mT1;
   mT0 = t0;
   mT1 = t1;

   // Apply effect

   // Effect is already inited; we call Process, End, and then Init
   // again, so the state is exactly the way it was before Preview
   // was called.
   mProgress = new ProgressDialog(StripAmpersand(GetEffectName()),
                                  _("Preparing preview"), 
                                  pdlgHideCancelButton); // Have only "Stop" button.
   bool bSuccess = Process();
   delete mProgress;
   End();
   Init();
   if (bSuccess)
   {
      mT0 = t0save;
      mT1 = t1save;

      WaveTrackArray playbackTracks;
      WaveTrackArray recordingTracks;
      // Probably not the same tracks post-processing, so can't rely on previous values of mixLeft & mixRight.
      TrackListOfKindIterator iter(Track::Wave, mTracks); 
      mixLeft = (WaveTrack*)(iter.First());
      mixRight = (WaveTrack*)(iter.Next());
      playbackTracks.Add(mixLeft);
      if (mixRight)
         playbackTracks.Add(mixRight);

#ifdef EXPERIMENTAL_MIDI_OUT
      NoteTrackArray empty;
#endif
      // Start audio playing
      int token =
         gAudioIO->StartStream(playbackTracks, recordingTracks, 
#ifdef EXPERIMENTAL_MIDI_OUT
                               empty,
#endif
                               NULL, rate, t0, t1, NULL);

      if (token) {
         int previewing = eProgressSuccess;

         mProgress = new ProgressDialog(StripAmpersand(GetEffectName()),
                                        _("Previewing"), pdlgHideCancelButton);

         while (gAudioIO->IsStreamActive(token) && previewing == eProgressSuccess) {
            ::wxMilliSleep(100);
            previewing = mProgress->Update(gAudioIO->GetStreamTime() - t0, t1 - t0);
//.........这里部分代码省略.........

bool ContrastDialog::GetDB(float &dB)
{
   float rms = float(0.0);
   int numberSelecteTracks = 0;

   // For stereo tracks: sqrt((mean(L)+mean(R))/2)
   bool isStereo = false;
   double meanSq = 0.0;

   AudacityProject *p = GetActiveProject();
   SelectedTrackListOfKindIterator iter(Track::Wave, p->GetTracks());
   WaveTrack *t = (WaveTrack *) iter.First();
   while (t) {
      numberSelecteTracks++;
      if (numberSelecteTracks > 1 && !isStereo) {
         AudacityMessageDialog m(NULL, _("You can only measure one track at a time."), _("Error"), wxOK);
         m.ShowModal();
         return false;
      }
      isStereo = t->GetLinked();

      wxASSERT(mT0 <= mT1);

      // Ignore whitespace beyond ends of track.
      if(mT0 < t->GetStartTime())
         mT0 = t->GetStartTime();
      if(mT1 > t->GetEndTime())
         mT1 = t->GetEndTime();

      auto SelT0 = t->TimeToLongSamples(mT0);
      auto SelT1 = t->TimeToLongSamples(mT1);

      if(SelT0 > SelT1)
      {
         AudacityMessageDialog m(NULL, _("Invalid audio selection.\nPlease ensure that audio is selected."), _("Error"), wxOK);
         m.ShowModal();
         return false;
      }

      if(SelT0 == SelT1)
      {
         AudacityMessageDialog m(NULL, _("Nothing to measure.\nPlease select a section of a track."), _("Error"), wxOK);
         m.ShowModal();
         return false;
      }

      // Don't throw in this analysis dialog
      rms = ((WaveTrack *)t)->GetRMS(mT0, mT1, false);
      meanSq += rms * rms;
      t = (WaveTrack *) iter.Next();
   }
   // TODO: This works for stereo, provided the audio clips are in both channels.
   // We should really count gaps between clips as silence.
   rms = (meanSq > 0.0)? sqrt(meanSq/(double)numberSelecteTracks) : 0.0;

   if(numberSelecteTracks == 0) {
      AudacityMessageDialog m(NULL, _("Please select an audio track."), _("Error"), wxOK);
      m.ShowModal();
      return false;
   }
   // Gives warning C4056, Overflow in floating-point constant arithmetic
   // -INFINITY is intentional here.
   // Looks like we are stuck with this warning, as 
   // #pragma warning( disable : 4056)
   // even around the whole function does not disable it successfully.

   dB = (rms == 0.0)? -INFINITY : LINEAR_TO_DB(rms);
   return true;
}

bool EffectTruncSilence::Process()
{
   SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
   WaveTrack *t;
   double t0 = mT0;
   double t1 = mT1;
   int tndx; 
   int tcount = 0;
   int fr;

   // Init using first track
   t = (WaveTrack *) iter.First();
   double rate = t->GetRate();
   sampleCount blockLen = t->GetMaxBlockSize();

   // Get the left and right bounds for all tracks
   while (t) {
      // Make sure all tracks have the same sample rate
      if (rate != t->GetRate()) {
         wxMessageBox(_("All tracks must have the same sample rate"), _("Truncate Silence"));
         return false;
      }

      // Count the tracks
      tcount++;

      // Set the current bounds to whichever left marker is
      // greater and whichever right marker is less
      t0 = wxMax(mT0, t->GetStartTime());
      t1 = wxMin(mT1, t->GetEndTime());

      // Use the smallest block size of all the tracks
      blockLen = wxMin(blockLen, t->GetMaxBlockSize());

      // Iterate to the next track
      t = (WaveTrack*) iter.Next();
   }

   // Just a sanity check, really it should be much higher
   if(blockLen < 4*mBlendFrameCount)
      blockLen = 4*mBlendFrameCount;

   // Transform the marker timepoints to samples
   t = (WaveTrack *) iter.First();
   sampleCount start = t->TimeToLongSamples(t0);
   sampleCount end = t->TimeToLongSamples(t1);

   // Bigger buffers reduce 'reset'
   //blockLen *= 8;
   // Stress-test the logic for cutting samples through block endpoints
   //blockLen /= 8;

   // Set thresholds
   // We have a lower bound on the amount of silence we chop out at a time
   // to avoid chopping up low frequency sounds.  We're good down to 10Hz
   // if we use 100ms.
   const float minTruncMs = 1.0f;
   double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];
   int truncInitialAllowedSilentSamples = 
      int((wxMax( mTruncInitialAllowedSilentMs, minTruncMs) * rate) / 1000.0);
   int truncLongestAllowedSilentSamples = 
      int((wxMax( mTruncLongestAllowedSilentMs, minTruncMs) * rate) / 1000.0);

   // Require at least 4 samples for lengths
   if(truncInitialAllowedSilentSamples < 4)
      truncInitialAllowedSilentSamples = 4;
   if(truncLongestAllowedSilentSamples < 4)
      truncLongestAllowedSilentSamples = 4;

   // If the cross-fade is longer than the minimum length,
   // then limit the cross-fade length to the minimum length
   // This allows us to have reasonable cross-fade by default
   // and still allow for 1ms minimum lengths
   if(truncInitialAllowedSilentSamples < mBlendFrameCount)
      mBlendFrameCount = truncInitialAllowedSilentSamples;
   if(truncLongestAllowedSilentSamples < mBlendFrameCount)
      mBlendFrameCount = truncLongestAllowedSilentSamples;

   // For sake of efficiency, don't let blockLen be less than double the longest silent samples
   // up until a sane limit of 1Meg samples
   while((blockLen > 0) && (blockLen < truncLongestAllowedSilentSamples*2) && (blockLen < 1048576)) {
      blockLen *= 2;
   }
    // Don't allow either value to be more than half of the block length
   if(truncLongestAllowedSilentSamples > blockLen/2)
      truncLongestAllowedSilentSamples = blockLen/2;
   if(truncInitialAllowedSilentSamples > truncLongestAllowedSilentSamples)
      truncInitialAllowedSilentSamples = truncLongestAllowedSilentSamples;

   // We use the 'longest' variable as additive to the 'initial' variable
   truncLongestAllowedSilentSamples -= truncInitialAllowedSilentSamples;

   // Perform the crossfade half-way through the minimum removed silence duration
   int rampInFrames = (truncInitialAllowedSilentSamples + mBlendFrameCount) / 2;
   if(rampInFrames > truncInitialAllowedSilentSamples)
      rampInFrames = truncInitialAllowedSilentSamples;

   // Allocate buffers
   float **buffer = new float*[tcount];
   for (tndx = 0; tndx < tcount; tndx++) {
//.........这里部分代码省略.........

bool EffectChangeSpeed::Process()
{
	// Similar to EffectSoundTouch::Process()

   //Iterate over each track
   this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
   bool bGoodResult = true;

   TrackListIterator iter(mOutputWaveTracks);
   WaveTrack* pOutWaveTrack = (WaveTrack*)(iter.First());
   mCurTrackNum = 0;
	m_maxNewLength = 0.0;
	
   //Get start and end times from track
   mCurT0 = pOutWaveTrack->GetStartTime();
   mCurT1 = pOutWaveTrack->GetEndTime();

   //Set the current bounds to whichever left marker is
   //greater and whichever right marker is less:
   mCurT0 = wxMax(mT0, mCurT0);
   mCurT1 = wxMin(mT1, mCurT1);

   double len = pOutWaveTrack->GetEndTime() - pOutWaveTrack->GetStartTime();
   
   while (pOutWaveTrack != NULL)
   {
      //Get start and end times from track
      mCurT0 = pOutWaveTrack->GetStartTime();
      mCurT1 = pOutWaveTrack->GetEndTime();

      //Set the current bounds to whichever left marker is
      //greater and whichever right marker is less:
      mCurT0 = wxMax(mT0, mCurT0);
      mCurT1 = wxMin(mT1, mCurT1);

      // Process only if the right marker is to the right of the left marker
      if (mCurT1 > mCurT0) {       
         //Transform the marker timepoints to samples
         sampleCount start = pOutWaveTrack->TimeToLongSamples(mCurT0);
         sampleCount end = pOutWaveTrack->TimeToLongSamples(mCurT1);

         //ProcessOne() (implemented below) processes a single track
         if (!ProcessOne(pOutWaveTrack, start, end))
         {
            bGoodResult = false;
            break;
         }
      }
      
      //Iterate to the next track
      pOutWaveTrack = (WaveTrack*)(iter.Next());
      mCurTrackNum++;
   }

   this->ReplaceProcessedWaveTracks(bGoodResult); 

#ifdef EXPERIMENTAL_FULL_LINKING
   AudacityProject *p = (AudacityProject*)mParent;
   if( p && p->IsSticky() ){
      pOutWaveTrack = (WaveTrack*)(iter.First());
      double newLen = pOutWaveTrack->GetEndTime() - pOutWaveTrack->GetStartTime();
      double timeAdded = newLen-len;
      double sel = mCurT1-mCurT0;
      double percent = (sel/(timeAdded+sel))*100 - 100;
      if ( !(HandleGroupChangeSpeed(percent, mCurT0, mCurT1)) ) bGoodResult = false;
   }
#endif

// mT1 = mT0 + m_maxNewLength; // Update selection.

   return bGoodResult;
}

// Deduce m_FromFrequency from the samples at the beginning of
// the selection. Then set some other params accordingly.
void EffectChangePitch::DeduceFrequencies()
{
   // As a neat trick, attempt to get the frequency of the note at the
   // beginning of the selection.
   SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
   WaveTrack *track = (WaveTrack *) iter.First();
   if (track) {
      double rate = track->GetRate();

      // Auto-size window -- high sample rates require larger windowSize.
      // Aim for around 2048 samples at 44.1 kHz (good down to about 100 Hz).
      // To detect single notes, analysis period should be about 0.2 seconds.
      // windowSize must be a power of 2.
      int windowSize = wxRound(pow(2.0, floor((log(rate / 20.0)/log(2.0)) + 0.5)));
      // windowSize < 256 too inaccurate
      windowSize = (windowSize > 256)? windowSize : 256;

      // we want about 0.2 seconds to catch the first note.
      // number of windows rounded to nearest integer >= 1.
      int numWindows = wxRound((double)(rate / (5.0f * windowSize)));
      numWindows = (numWindows > 0)? numWindows : 1;

      double trackStart = track->GetStartTime();
      double t0 = mT0 < trackStart? trackStart: mT0;
      sampleCount start = track->TimeToLongSamples(t0);

      int analyzeSize = windowSize * numWindows;
      float * buffer;
      buffer = new float[analyzeSize];

      float * freq;
      freq = new float[windowSize/2];

      float * freqa;
      freqa = new float[windowSize/2];

      int i, j, argmax;
      int lag;

      for(j=0; j<windowSize/2; j++)
         freqa[j] = 0;

      track->Get((samplePtr) buffer, floatSample, start, analyzeSize);
      for(i=0; i<numWindows; i++) {
         ComputeSpectrum(buffer+i*windowSize, windowSize,
                         windowSize, rate, freq, true);
         for(j=0; j<windowSize/2; j++)
            freqa[j] += freq[j];
      }
      argmax=0;
      for(j=1; j<windowSize/2; j++)
         if (freqa[j] > freqa[argmax])
            argmax = j;

      delete [] freq;
      delete [] freqa;
      delete [] buffer;

      lag = (windowSize/2 - 1) - argmax;
      m_dStartFrequency = rate / lag;
   }

   double dFromMIDInote = FreqToMIDInote(m_dStartFrequency);
   double dToMIDInote = dFromMIDInote + m_dSemitonesChange;
   m_nFromPitch = PitchIndex(dFromMIDInote);
   m_nFromOctave = PitchOctave(dFromMIDInote);
   m_nToPitch = PitchIndex(dToMIDInote);
   m_nToOctave = PitchOctave(dToMIDInote);

   m_FromFrequency = m_dStartFrequency;
   Calc_PercentChange();
   Calc_ToFrequency();
}

bool EffectRepair::Process()
{
   //v This may be too much copying for EffectRepair. To support Cancel, may be able to copy much less.
   //  But for now, Cancel isn't supported without this.
   this->CopyInputTracks(); // Set up mOutputTracks. //v This may be too much copying for EffectRepair.
   bool bGoodResult = true;

   SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks.get());
   WaveTrack *track = (WaveTrack *) iter.First();
   int count = 0;
   while (track) {
      const
      double trackStart = track->GetStartTime();
      const double repair_t0 = std::max(mT0, trackStart);
      const
      double trackEnd = track->GetEndTime();
      const double repair_t1 = std::min(mT1, trackEnd);
      const
      double repair_deltat = repair_t1 - repair_t0;
      if (repair_deltat > 0) {  // selection is within track audio
         const auto repair0 = track->TimeToLongSamples(repair_t0);
         const auto repair1 = track->TimeToLongSamples(repair_t1);
         const auto repairLen = repair1 - repair0;
         if (repairLen > 128) {
            ::Effect::MessageBox(_("The Repair effect is intended to be used on very short sections of damaged audio (up to 128 samples).\n\nZoom in and select a tiny fraction of a second to repair."));
            bGoodResult = false;
            break;
         }

         const double rate = track->GetRate();
         const double spacing = std::max(repair_deltat * 2, 128. / rate);
         const double t0 = std::max(repair_t0 - spacing, trackStart);
         const double t1 = std::min(repair_t1 + spacing, trackEnd);

         const auto s0 = track->TimeToLongSamples(t0);
         const auto s1 = track->TimeToLongSamples(t1);
         // The difference is at most 2 * 128:
         const auto repairStart = (repair0 - s0).as_size_t();
         const auto len = s1 - s0;

         if (s0 == repair0 && s1 == repair1) {
            ::Effect::MessageBox(_("Repair works by using audio data outside the selection region.\n\nPlease select a region that has audio touching at least one side of it.\n\nThe more surrounding audio, the better it performs."));
   ///            The Repair effect needs some data to go on.\n\nPlease select an area to repair with some audio on at least one side (the more the better)."));
            bGoodResult = false;
            break;
         }

         if (!ProcessOne(count, track, s0,
                         // len is at most 5 * 128.
                         len.as_size_t(),
                         repairStart,
                         // repairLen is at most 128.
                         repairLen.as_size_t() )) {
            bGoodResult = false;
            break;
         }
      }

      track = (WaveTrack *) iter.Next();
      count++;
   }

   this->ReplaceProcessedTracks(bGoodResult);
   return bGoodResult;
}

bool EffectTruncSilence::Process()
{
   TrackListIterator iter(mWaveTracks);
   WaveTrack *t;
   double t0 = mT0;
   double t1 = mT1;
   int tndx; 
   int tcount = 0;

   // Init using first track
   t = (WaveTrack *) iter.First();
   double rate = t->GetRate();
   sampleCount blockLen = t->GetMaxBlockSize();

   // Get the left and right bounds for all tracks
   while (t) {
      // Make sure all tracks have the same sample rate
      if (rate != t->GetRate()) {
         wxMessageBox(_("All tracks must have the same sample rate"), _("Truncate Silence"));
         return false;
      }

      // Count the tracks
      tcount++;

      // Set the current bounds to whichever left marker is
      // greater and whichever right marker is less
      t0 = wxMax(mT0, t->GetStartTime());
      t1 = wxMin(mT1, t->GetEndTime());

      // Use the smallest block size of all the tracks
      blockLen = wxMin(blockLen, t->GetMaxBlockSize());

      // Iterate to the next track
      t = (WaveTrack*) iter.Next();
   }

   // Transform the marker timepoints to samples
   t = (WaveTrack *) iter.First();
   longSampleCount start = t->TimeToLongSamples(t0);
   longSampleCount end = t->TimeToLongSamples(t1);

   // Bigger buffers reduce 'reset'
   blockLen *= 8;

   // Allocate buffers
   float **buffer = new float*[tcount];
   for (tndx = 0; tndx < tcount; tndx++) {
      buffer[tndx] = new float[blockLen];
   }

   // Set thresholds
   // We have a lower bound on the amount of silence we chop out at a time
   // to avoid chopping up low frequency sounds.  We're good down to 10Hz
   // if we use 100ms.
   const float minTruncMs = 1.0f;
   double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];
   int truncLongestAllowedSilentSamples = 
      int((wxMax( mTruncLongestAllowedSilentMs, minTruncMs) * rate) / 1000.0);

   

   // Figure out number of frames for ramping
   int quarterSecondFrames = int((rate * QUARTER_SECOND_MS) / 1000.0);
   int rampInFrames = (truncLongestAllowedSilentSamples / 4);
   if (rampInFrames > quarterSecondFrames) {
      rampInFrames = quarterSecondFrames;
   }

   // Start processing
   this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
   TrackListIterator iterOut(mOutputWaveTracks);

   longSampleCount index = start;
   longSampleCount outTrackOffset = start;
   bool cancelled = false;
   while (index < end) {

      // Limit size of current block if we've reached the end
      sampleCount limit = blockLen;
      if ((index + blockLen) > end) {
         limit = end - index; 
      }

      // Fill the buffers
      tndx = 0;
      t = (WaveTrack *) iter.First();
      while (t) {
         t->Get((samplePtr)buffer[tndx++], floatSample, index, blockLen);
         t = (WaveTrack *) iter.Next();
      }

      // Reset
      bool ignoringFrames = false;
      sampleCount consecutiveSilentFrames = 0;
      sampleCount truncIndex = 0;

      // Look for silences in current block
      for (sampleCount i = 0; i < limit; i++) {

//.........这里部分代码省略.........

bool EffectSBSMS::Process()
{
   bool bGoodResult = true;

   //Iterate over each track
   //Track::All is needed because this effect needs to introduce silence in the group tracks to keep sync
   this->CopyInputTracks(Track::All); // Set up mOutputTracks.
   TrackListIterator iter(mOutputTracks);
   Track* t;
   mCurTrackNum = 0;

   double maxDuration = 0.0;

   // Must sync if selection length will change
   bool mustSync = (rateStart != rateEnd);
   Slide rateSlide(rateSlideType,rateStart,rateEnd);
   Slide pitchSlide(pitchSlideType,pitchStart,pitchEnd);
   mTotalStretch = rateSlide.getTotalStretch();

   t = iter.First();
   while (t != NULL) {
      if (t->GetKind() == Track::Label &&
            (t->GetSelected() || (mustSync && t->IsSyncLockSelected())) )
      {
         if (!ProcessLabelTrack(t)) {
            bGoodResult = false;
            break;
         }
      }
      else if (t->GetKind() == Track::Wave && t->GetSelected() )
      {
         WaveTrack* leftTrack = (WaveTrack*)t;

         //Get start and end times from track
         mCurT0 = leftTrack->GetStartTime();
         mCurT1 = leftTrack->GetEndTime();

         //Set the current bounds to whichever left marker is
         //greater and whichever right marker is less
         mCurT0 = wxMax(mT0, mCurT0);
         mCurT1 = wxMin(mT1, mCurT1);

         // Process only if the right marker is to the right of the left marker
         if (mCurT1 > mCurT0) {
            sampleCount start;
            sampleCount end;
            start = leftTrack->TimeToLongSamples(mCurT0);
            end = leftTrack->TimeToLongSamples(mCurT1);

            WaveTrack* rightTrack = NULL;
            if (leftTrack->GetLinked()) {
               double t;
               rightTrack = (WaveTrack*)(iter.Next());

               //Adjust bounds by the right tracks markers
               t = rightTrack->GetStartTime();
               t = wxMax(mT0, t);
               mCurT0 = wxMin(mCurT0, t);
               t = rightTrack->GetEndTime();
               t = wxMin(mT1, t);
               mCurT1 = wxMax(mCurT1, t);

               //Transform the marker timepoints to samples
               start = leftTrack->TimeToLongSamples(mCurT0);
               end = leftTrack->TimeToLongSamples(mCurT1);

               mCurTrackNum++; // Increment for rightTrack, too.
            }
            sampleCount trackStart = leftTrack->TimeToLongSamples(leftTrack->GetStartTime());
            sampleCount trackEnd = leftTrack->TimeToLongSamples(leftTrack->GetEndTime());

            // SBSMS has a fixed sample rate - we just convert to its sample rate and then convert back
            float srTrack = leftTrack->GetRate();
            float srProcess = bLinkRatePitch?srTrack:44100.0;

            // the resampler needs a callback to supply its samples
            ResampleBuf rb;
            sampleCount maxBlockSize = leftTrack->GetMaxBlockSize();
            rb.blockSize = maxBlockSize;
            rb.buf = (audio*)calloc(rb.blockSize,sizeof(audio));
            rb.leftTrack = leftTrack;
            rb.rightTrack = rightTrack?rightTrack:leftTrack;
            rb.leftBuffer = (float*)calloc(maxBlockSize,sizeof(float));
            rb.rightBuffer = (float*)calloc(maxBlockSize,sizeof(float));

            // Samples in selection
            sampleCount samplesIn = end-start;

            // Samples for SBSMS to process after resampling
            sampleCount samplesToProcess = (sampleCount) ((float)samplesIn*(srProcess/srTrack));

            SlideType outSlideType;
            SBSMSResampleCB outResampleCB;

            sampleCount processPresamples = 0;
            sampleCount trackPresamples = 0;

            if(bLinkRatePitch) {
              rb.bPitch = true;
              outSlideType = rateSlideType;
//.........这里部分代码省略.........

bool EffectRepair::Process()
{
   //v This may be too much copying for EffectRepair. To support Cancel, may be able to copy much less.
   //  But for now, Cancel isn't supported without this.
   this->CopyInputTracks(); // Set up mOutputTracks. //v This may be too much copying for EffectRepair.
   bool bGoodResult = true;

   SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks);
   WaveTrack *track = (WaveTrack *) iter.First();
   int count = 0;
   while (track) {
      double trackStart = track->GetStartTime();
      double trackEnd = track->GetEndTime();
      double repair_t0 = mT0;
      double repair_t1 = mT1;
      repair_t0 = (repair_t0 < trackStart? trackStart: repair_t0);
      repair_t1 = (repair_t1 > trackEnd? trackEnd: repair_t1);
      if (repair_t0 < repair_t1) {  // selection is within track audio
         double rate = track->GetRate();
         double repair_deltat = repair_t1 - repair_t0;

         double spacing = repair_deltat * 2;

         if (spacing < 128. / rate)
            spacing = 128. / rate;

         double t0 = repair_t0 - spacing;
         double t1 = repair_t1 + spacing;

         t0 = t0 < trackStart? trackStart: t0;
         t1 = t1 > trackEnd? trackEnd: t1;

         repair_t0 = (repair_t0 < t0? t0: repair_t0);
         repair_t1 = (repair_t1 > t1? t1: repair_t1);

         sampleCount s0 = track->TimeToLongSamples(t0);
         sampleCount repair0 = track->TimeToLongSamples(repair_t0);
         sampleCount repair1 = track->TimeToLongSamples(repair_t1);
         sampleCount s1 = track->TimeToLongSamples(t1);

         sampleCount repairStart = (sampleCount)(repair0 - s0);
         sampleCount repairLen = (sampleCount)(repair1 - repair0);
         sampleCount len = (sampleCount)(s1 - s0);

         if (repairLen > 128) {
            ::wxMessageBox(_("The Repair effect is intended to be used on very short sections of damaged audio (up to 128 samples).\n\nZoom in and select a tiny fraction of a second to repair."));
            bGoodResult = false;
            break;
         }

         if (s0 == repair0 && s1 == repair1) {
            ::wxMessageBox(_("Repair works by using audio data outside the selection region.\n\nPlease select a region that has audio touching at least one side of it.\n\nThe more surrounding audio, the better it performs."));
   ///            The Repair effect needs some data to go on.\n\nPlease select an area to repair with some audio on at least one side (the more the better)."));
            bGoodResult = false;
            break;
         }

         if (!ProcessOne(count, track,
                         s0, len, repairStart, repairLen)) {
            bGoodResult = false;
            break;
         }
      }

      track = (WaveTrack *) iter.Next();
      count++;
   }

   this->ReplaceProcessedTracks(bGoodResult);
   return bGoodResult;
}

bool EffectSoundTouch::Process()
{
   // Assumes that mSoundTouch has already been initialized
   // by the subclass for subclass-specific parameters. The
   // time warper should also be set.

   // Check if this effect will alter the selection length; if so, we need
   // to operate on sync-lock selected tracks.
   bool mustSync = true;
   if (mT1 == GetTimeWarper()->Warp(mT1)) {
      mustSync = false;
   }

   //Iterate over each track
   // Needs Track::All for sync-lock grouping.
   this->CopyInputTracks(Track::All);
   bool bGoodResult = true;

   TrackListIterator iter(mOutputTracks);
   Track* t;
   mCurTrackNum = 0;
   m_maxNewLength = 0.0;

   t = iter.First();
   while (t != NULL) {
      if (t->GetKind() == Track::Label &&
            (t->GetSelected() || (mustSync && t->IsSyncLockSelected())) )
      {
         if (!ProcessLabelTrack(t))
         {
            bGoodResult = false;
            break;
         }
      }
#ifdef USE_MIDI
      else if (t->GetKind() == Track::Note &&
               (t->GetSelected() || (mustSync && t->IsSyncLockSelected())))
      {
         if (!ProcessNoteTrack(t))
         {
            bGoodResult = false;
            break;
         }
      }
#endif
      else if (t->GetKind() == Track::Wave && t->GetSelected())
      {
         WaveTrack* leftTrack = (WaveTrack*)t;
         //Get start and end times from track
         mCurT0 = leftTrack->GetStartTime();
         mCurT1 = leftTrack->GetEndTime();

         //Set the current bounds to whichever left marker is
         //greater and whichever right marker is less
         mCurT0 = wxMax(mT0, mCurT0);
         mCurT1 = wxMin(mT1, mCurT1);

         // Process only if the right marker is to the right of the left marker
         if (mCurT1 > mCurT0) {
            sampleCount start, end;

            if (leftTrack->GetLinked()) {
               double t;
               WaveTrack* rightTrack = (WaveTrack*)(iter.Next());

               //Adjust bounds by the right tracks markers
               t = rightTrack->GetStartTime();
               t = wxMax(mT0, t);
               mCurT0 = wxMin(mCurT0, t);
               t = rightTrack->GetEndTime();
               t = wxMin(mT1, t);
               mCurT1 = wxMax(mCurT1, t);

               //Transform the marker timepoints to samples
               start = leftTrack->TimeToLongSamples(mCurT0);
               end = leftTrack->TimeToLongSamples(mCurT1);

               //Inform soundtouch there's 2 channels
               mSoundTouch->setChannels(2);

               //ProcessStereo() (implemented below) processes a stereo track
               if (!ProcessStereo(leftTrack, rightTrack, start, end))
               {
                  bGoodResult = false;
                  break;
               }
               mCurTrackNum++; // Increment for rightTrack, too.
            } else {
               //Transform the marker timepoints to samples
               start = leftTrack->TimeToLongSamples(mCurT0);
               end = leftTrack->TimeToLongSamples(mCurT1);

               //Inform soundtouch there's a single channel
               mSoundTouch->setChannels(1);

               //ProcessOne() (implemented below) processes a single track
               if (!ProcessOne(leftTrack, start, end))
               {
                  bGoodResult = false;
                  break;
//.........这里部分代码省略.........

bool EffectSBSMS::Process()
{
   if(!bInit) {
      sbsms_init(4096);
      bInit = TRUE;
   }
   
   bool bGoodResult = true;
   
   //Iterate over each track
   //Track::All is needed because this effect needs to introduce silence in the group tracks to keep sync
   this->CopyInputTracks(Track::All); // Set up mOutputTracks.
   TrackListIterator iter(mOutputTracks);
   Track* t;
   mCurTrackNum = 0;

   double maxDuration = 0.0;

   if(rateStart == rateEnd)
      mTotalStretch = 1.0/rateStart;
   else
      mTotalStretch = 1.0/(rateEnd-rateStart)*log(rateEnd/rateStart);

   // Must sync if selection length will change
   bool mustSync = (mTotalStretch != 1.0);

   t = iter.First();
   while (t != NULL) {
      if (t->GetKind() == Track::Label && 
            (t->GetSelected() || (mustSync && t->IsSynchroSelected())) )
      {
         if (!ProcessLabelTrack(t)) {
            bGoodResult = false;
            break;
         }
      }
      else if (t->GetKind() == Track::Wave && t->GetSelected() )
      {
         WaveTrack* leftTrack = (WaveTrack*)t;

         //Get start and end times from track
         mCurT0 = leftTrack->GetStartTime();
         mCurT1 = leftTrack->GetEndTime();
         
         //Set the current bounds to whichever left marker is
         //greater and whichever right marker is less
         mCurT0 = wxMax(mT0, mCurT0);
         mCurT1 = wxMin(mT1, mCurT1);
         
         // Process only if the right marker is to the right of the left marker
         if (mCurT1 > mCurT0) {
            sampleCount start;
            sampleCount end;
            start = leftTrack->TimeToLongSamples(mCurT0);
            end = leftTrack->TimeToLongSamples(mCurT1);
            
            WaveTrack* rightTrack = NULL;
            if (leftTrack->GetLinked()) {
               double t;
               rightTrack = (WaveTrack*)(iter.Next());
               
               //Adjust bounds by the right tracks markers
               t = rightTrack->GetStartTime();
               t = wxMax(mT0, t);
               mCurT0 = wxMin(mCurT0, t);
               t = rightTrack->GetEndTime();
               t = wxMin(mT1, t);
               mCurT1 = wxMax(mCurT1, t);
               
               //Transform the marker timepoints to samples
               start = leftTrack->TimeToLongSamples(mCurT0);
               end = leftTrack->TimeToLongSamples(mCurT1);
               
               mCurTrackNum++; // Increment for rightTrack, too.	
            }
            
            sampleCount trackEnd = leftTrack->TimeToLongSamples(leftTrack->GetEndTime());

            // SBSMS has a fixed sample rate - we just convert to its sample rate and then convert back
            float srIn = leftTrack->GetRate();
            float srSBSMS = 44100.0;
            
            // the resampler needs a callback to supply its samples
            resampleBuf rb;
            sampleCount maxBlockSize = leftTrack->GetMaxBlockSize();
            rb.block = maxBlockSize;
            rb.buf = (audio*)calloc(rb.block,sizeof(audio));
            rb.leftTrack = leftTrack;
            rb.rightTrack = rightTrack?rightTrack:leftTrack;
            rb.leftBuffer = (float*)calloc(maxBlockSize,sizeof(float));
            rb.rightBuffer = (float*)calloc(maxBlockSize,sizeof(float));
            rb.offset = start;
            rb.end = trackEnd;
            rb.ratio = srSBSMS/srIn;
            rb.resampler = new Resampler(resampleCB, &rb);
            
            // Samples in selection
            sampleCount samplesIn = end-start;
            
            // Samples for SBSMS to process after resampling
//.........这里部分代码省略.........

bool EffectChangeSpeed::Process()
{
   // Similar to EffectSoundTouch::Process()

   // Iterate over each track.
   // Track::All is needed because this effect needs to introduce 
   // silence in the sync-lock group tracks to keep sync
   this->CopyInputTracks(Track::All); // Set up mOutputTracks.
   bool bGoodResult = true;

   TrackListIterator iter(mOutputTracks);
   Track* t;
   mCurTrackNum = 0;
   mMaxNewLength = 0.0;

   mFactor = 100.0 / (100.0 + mPercentChange);

   t = iter.First();
   while (t != NULL)
   {
      if (t->GetKind() == Track::Label) {
         if (t->GetSelected() || t->IsSyncLockSelected())
         {
            if (!ProcessLabelTrack(t)) {
               bGoodResult = false;
               break;
            }
         }
      }
      else if (t->GetKind() == Track::Wave && t->GetSelected())
      {
         WaveTrack *pOutWaveTrack = (WaveTrack*)t;
         //Get start and end times from track
         mCurT0 = pOutWaveTrack->GetStartTime();
         mCurT1 = pOutWaveTrack->GetEndTime();

         //Set the current bounds to whichever left marker is
         //greater and whichever right marker is less:
         mCurT0 = wxMax(mT0, mCurT0);
         mCurT1 = wxMin(mT1, mCurT1);

         // Process only if the right marker is to the right of the left marker
         if (mCurT1 > mCurT0) {
            //Transform the marker timepoints to samples
            sampleCount start = pOutWaveTrack->TimeToLongSamples(mCurT0);
            sampleCount end = pOutWaveTrack->TimeToLongSamples(mCurT1);

            //ProcessOne() (implemented below) processes a single track
            if (!ProcessOne(pOutWaveTrack, start, end))
            {
               bGoodResult = false;
               break;
            }
         }
         mCurTrackNum++;
      }
      else if (t->IsSyncLockSelected())
      {
         t->SyncLockAdjust(mT1, mT0 + (mT1 - mT0) * mFactor);
      }

      //Iterate to the next track
      t=iter.Next();
   }

   if (bGoodResult)
      ReplaceProcessedTracks(bGoodResult);

   mT1 = mT0 + mMaxNewLength; // Update selection.

   return bGoodResult;
}

bool ContrastDialog::GetDB(float &dB)
{
   float rms = float(0.0);
   int numberSelecteTracks = 0;

   // For stereo tracks: sqrt((mean(L)+mean(R))/2)
   bool isStereo = false;
   double meanSq = 0.0;

   AudacityProject *p = GetActiveProject();
   SelectedTrackListOfKindIterator iter(Track::Wave, p->GetTracks());
   WaveTrack *t = (WaveTrack *) iter.First();
   while (t) {
      numberSelecteTracks++;
      if (numberSelecteTracks > 1 && !isStereo) {
         wxMessageDialog m(NULL, _("You can only measure one track at a time."), _("Error"), wxOK);
         m.ShowModal();
         return false;
      }
      isStereo = t->GetLinked();

      wxASSERT(mT0 <= mT1);

      // Ignore whitespace beyond ends of track.
      if(mT0 < t->GetStartTime())
         mT0 = t->GetStartTime();
      if(mT1 > t->GetEndTime())
         mT1 = t->GetEndTime();

      sampleCount SelT0 = t->TimeToLongSamples(mT0);
      sampleCount SelT1 = t->TimeToLongSamples(mT1);

      if(SelT0 > SelT1)
      {
         wxMessageDialog m(NULL, _("Invalid audio selection.\nPlease ensure that audio is selected."), _("Error"), wxOK);
         m.ShowModal();
         return false;
      }

      if(SelT0 == SelT1)
      {
         wxMessageDialog m(NULL, _("Nothing to measure.\nPlease select a section of a track."), _("Error"), wxOK);
         m.ShowModal();
         return false;
      }

      ((WaveTrack *)t)->GetRMS(&rms, mT0, mT1);
      meanSq += rms * rms;
      t = (WaveTrack *) iter.Next();
   }
   // TODO: This works for stereo, provided the audio clips are in both channels.
   // We should really count gaps between clips as silence.
   rms = (meanSq > 0.0)? sqrt(meanSq/(double)numberSelecteTracks) : 0.0;

   if(numberSelecteTracks == 0) {
      wxMessageDialog m(NULL, _("Please select an audio track."), _("Error"), wxOK);
      m.ShowModal();
      return false;
   }

   dB = (rms == 0.0)? -INFINITY : LINEAR_TO_DB(rms);
   return true;
}