C++ WaveTrack::Get方法代码示例

void FreqWindow::GetAudio()
{
   mData.reset();
   mDataLen = 0;

   int selcount = 0;
   bool warning = false;
   TrackListIterator iter(p->GetTracks());
   Track *t = iter.First();
   while (t) {
      if (t->GetSelected() && t->GetKind() == Track::Wave) {
         WaveTrack *track = (WaveTrack *)t;
         if (selcount==0) {
            mRate = track->GetRate();
            auto start = track->TimeToLongSamples(p->mViewInfo.selectedRegion.t0());
            auto end = track->TimeToLongSamples(p->mViewInfo.selectedRegion.t1());
            auto dataLen = end - start;
            if (dataLen > 10485760) {
               warning = true;
               mDataLen = 10485760;
            }
            else
               // dataLen is not more than 10 * 2 ^ 20
               mDataLen = dataLen.as_size_t();
            mData = Floats{ mDataLen };
            // Don't allow throw for bad reads
            track->Get((samplePtr)mData.get(), floatSample, start, mDataLen,
                       fillZero, false);
         }
         else {
            if (track->GetRate() != mRate) {
               AudacityMessageBox(_("To plot the spectrum, all selected tracks must be the same sample rate."));
               mData.reset();
               mDataLen = 0;
               return;
            }
            auto start = track->TimeToLongSamples(p->mViewInfo.selectedRegion.t0());
            Floats buffer2{ mDataLen };
            // Again, stop exceptions
            track->Get((samplePtr)buffer2.get(), floatSample, start, mDataLen,
                       fillZero, false);
            for (size_t i = 0; i < mDataLen; i++)
               mData[i] += buffer2[i];
         }
         selcount++;
      }
      t = iter.Next();
   }

   if (selcount == 0)
      return;

   if (warning) {
      wxString msg;
      msg.Printf(_("Too much audio was selected.  Only the first %.1f seconds of audio will be analyzed."),
                          (mDataLen / mRate));
      AudacityMessageBox(msg);
   }
}

//This might continue over a number of blocks.
double VoiceKey::TestEnergy (WaveTrack & t, sampleCount start, sampleCount len)
{

   double sum = 1;
   auto s = start;                                //Keep track of start
   auto originalLen = len;                        //Keep track of the length of block to process (its not the length of t)
   const auto blockSize = limitSampleBufferSize(
      t.GetMaxBlockSize(), len);               //Determine size of sampling buffer
   float *buffer = new float[blockSize];       //Get a sampling buffer

   while(len > 0)
      {
         //Figure out how much to grab
         auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );

         t.Get((samplePtr)buffer,floatSample, s,block);                      //grab the block;

         //Now, go through the block and calculate energy
         for(decltype(block) i = 0; i< block; i++)
            {
               sum += buffer[i]*buffer[i];
            }

         len -= block;
         s += block;
      }

   delete [] buffer;
   return sum / originalLen.as_double();
}

// 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, inputTracks());
   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.
      const size_t windowSize =
         // windowSize < 256 too inaccurate
         std::max(256, wxRound(pow(2.0, floor((log(rate / 20.0)/log(2.0)) + 0.5))));

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

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

      auto analyzeSize = windowSize * numWindows;
      Floats buffer{ analyzeSize };

      Floats freq{ windowSize / 2 };
      Floats freqa{ windowSize / 2, true };

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

      auto 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();
}

void AudacityProject::OnPlotSpectrum(wxCommandEvent & event)
{
   int selcount = 0;
   WaveTrack *selt = NULL;
   TrackListIterator iter(mTracks);
   VTrack *t = iter.First();
   while (t) {
      if (t->GetSelected())
         selcount++;
      if (t->GetKind() == VTrack::Wave)
         selt = (WaveTrack *) t;
      t = iter.Next();
   }
   if (selcount != 1) {
      wxMessageBox(_("Please select a single track first.\n"));
      return;
   }

   /* This shouldn't be possible, since the menu is grayed out.
    * But we'll check just in case it does happen, to prevent
    * the crash that would result. */

   if (!selt) {
      wxMessageBox(_("Please select a track first.\n"));
      return;
   }


   sampleCount s0 = (sampleCount) ((mViewInfo.sel0 - selt->GetOffset())
                                   * selt->GetRate());
   sampleCount s1 = (sampleCount) ((mViewInfo.sel1 - selt->GetOffset())
                                   * selt->GetRate());
   sampleCount slen = s1 - s0;

   if (slen > 1048576)
      slen = 1048576;

   float *data = new float[slen];
   sampleType *data_sample = new sampleType[slen];

   if (s0 >= selt->GetNumSamples() || s0 + slen > selt->GetNumSamples()) {
      wxMessageBox(_("Not enough samples selected.\n"));
      delete[]data;
      delete[]data_sample;
      return;
   }

   selt->Get(data_sample, s0, slen);

   for (sampleCount i = 0; i < slen; i++)
      data[i] = data_sample[i] / 32767.;

   gFreqWindow->Plot(slen, data, selt->GetRate());
   gFreqWindow->Show(true);
   gFreqWindow->Raise();

   delete[]data;
   delete[]data_sample;
}

// DeduceFrequencies is Dominic's extremely cool trick (Vaughan sez so!) 
// to set deduce m_FromFrequency from the samples at the beginning of 
// the selection. Then we 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) {
      const int windowSize = 1024;
      const int analyzeSize = 8192;
      const int numWindows = analyzeSize / windowSize;
      double trackStart = track->GetStartTime();
      double t0 = mT0 < trackStart? trackStart: mT0;
      sampleCount start = track->TimeToLongSamples(t0);
      double rate = track->GetRate();
      float buffer[analyzeSize];
      float freq[windowSize/2];
      float freqa[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;
      lag = (windowSize/2 - 1) - argmax;
      m_FromFrequency = rate / lag;
      m_ToFrequency = (m_FromFrequency * (100.0 + m_PercentChange)) / 100.0;

      // Now we can set the pitch control values. 
      m_FromPitchIndex = PitchIndex(FreqToMIDInoteNumber(m_FromFrequency));
      m_bWantPitchDown = (m_ToFrequency < m_FromFrequency);
      m_ToPitchIndex = PitchIndex(FreqToMIDInoteNumber(m_ToFrequency));
   }
}

double VoiceKey::TestDirectionChanges(
   const WaveTrack & t, sampleCount start, sampleCount len)
{


   auto s = start;                                //Keep track of start
   auto originalLen = len;                        //Keep track of the length of block to process (its not the length of t)
   const auto blockSize = limitSampleBufferSize(
      t.GetMaxBlockSize(), len);               //Determine size of sampling buffer
   unsigned long directionchanges = 1;
   float lastval=float(0);
   int lastdirection=1;

   Floats buffer{ blockSize };       //Get a sampling buffer

   while(len > 0) {
      //Figure out how much to grab
      auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );

      t.Get((samplePtr)buffer.get(), floatSample, s, block);                      //grab the block;

      if  (len == originalLen) {
         //The first time through, set stuff up special.
         lastval = buffer[0];
      }

      //Now, go through the block and calculate zero crossings


      for(decltype(block) i = 0; i< block; i++){

         if( sgn(buffer[i]-lastval) != lastdirection) {
            directionchanges++;
            lastdirection = sgn(buffer[i] - lastval);
         }
         lastval = buffer[i];

      }
      len -= block;
      s += block;
   }
   return (double)directionchanges/originalLen.as_double();
}

double VoiceKey::TestSignChanges(WaveTrack & t, sampleCount start, sampleCount len)
{


   auto s = start;                                //Keep track of start
   auto originalLen = len;                        //Keep track of the length of block to process (its not the length of t)
   const auto blockSize = limitSampleBufferSize(
      t.GetMaxBlockSize(), len);               //Determine size of sampling buffer
   unsigned long signchanges = 1;
   int currentsign=0;

   float *buffer = new float[blockSize];       //Get a sampling buffer

   while(len > 0) {
      //Figure out how much to grab
      auto block = limitSampleBufferSize ( t.GetBestBlockSize(s), len );

      t.Get((samplePtr)buffer,floatSample, s,block);                      //grab the block;

      if  (len == originalLen)
         {
            //The first time through, set stuff up special.
            currentsign = sgn(buffer[0]);
         }

      //Now, go through the block and calculate zero crossings

      for(decltype(block) i = 0; i< block; i++)
         {
            if( sgn(buffer[i]) != currentsign)
               {
                  currentsign = sgn(buffer[i]);
                  signchanges++;
               }

         }
      len -= block;
      s += block;
   }
   delete [] buffer;
   return (double)signchanges / originalLen.as_double();
}

bool VampEffect::Process()
{
   if (!mPlugin) return false;

   TrackListIterator iter(mWaveTracks);

   int count = 0;

   WaveTrack *left = (WaveTrack *)iter.First();

   bool multiple = false;
   int prevTrackChannels = 0;

   TrackListIterator scooter(iter);
   if (left->GetLinked()) scooter.Next();      
   if (scooter.Next()) {
      // if there is another track beyond this one and any linked one,
      // then we're processing more than one track.  That means we
      // should use the originating track name in each new label
      // track's name, to make clear which is which
      multiple = true;
   }

   while (left) {

      sampleCount lstart, rstart;
      sampleCount len;
      GetSamples(left, &lstart, &len);
      
      WaveTrack *right = NULL;
      int channels = 1;

      if (left->GetLinked()) {
         right = (WaveTrack *)iter.Next();
         channels = 2;
         GetSamples(right, &rstart, &len);
      }

      size_t step = mPlugin->getPreferredStepSize();
      size_t block = mPlugin->getPreferredBlockSize();

      bool initialiseRequired = true;

      if (block == 0) {
         if (step != 0) block = step;
         else block = 1024;
      }
      if (step == 0) {
         step = block;
      }

      if (prevTrackChannels > 0) {
         // Plugin has already been initialised, so if the number of
         // channels remains the same, we only need to do a reset.
         // Otherwise we need to re-construct the whole plugin,
         // because a Vamp plugin can't be re-initialised.
         if (prevTrackChannels == channels) {
            mPlugin->reset();
            initialiseRequired = false;
         } else {
            //!!! todo: retain parameters previously set
            Init();
         }
      }

      if (initialiseRequired) {
         if (!mPlugin->initialise(channels, step, block)) {
            wxMessageBox(_("Sorry, Vamp Plug-in failed to initialize."));
            return false;
         }
      }

      LabelTrack *ltrack = mFactory->NewLabelTrack();

      if (!multiple) {
         ltrack->SetName(GetEffectName());
      } else {
         ltrack->SetName(wxString::Format(wxT("%s: %s"),
                                          left->GetName().c_str(),
                                          GetEffectName().c_str()));
      }

      mTracks->Add(ltrack);

      float **data = new float*[channels];
      for (int c = 0; c < channels; ++c) data[c] = new float[block];

      sampleCount originalLen = len;
      sampleCount ls = lstart;
      sampleCount rs = rstart;

      while (len) {
         
         int request = block;
         if (request > len) request = len;

         if (left) left->Get((samplePtr)data[0], floatSample, ls, request);
         if (right) right->Get((samplePtr)data[1], floatSample, rs, request);

         if (request < (int)block) {
//.........这里部分代码省略.........

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.........

//Move forward to find an ON region.
sampleCount VoiceKey::OnForward (WaveTrack & t, sampleCount start, sampleCount len)
{

   if((mWindowSize) >= (len + 10).as_double() ){

      /* i18n-hint: Voice key is an experimental/incomplete feature that
         is used to navigate in vocal recordings, to move forwards and
         backwards by words.  So 'key' is being used in the sense of an index.
         This error message means that you've selected too short
         a region of audio to be able to use this feature.*/
      wxMessageBox(_("Selection is too small to use voice key."));
      return start;
   }
   else {

      //Change the millisecond-based parameters into sample-based parameters
      double rate = t.GetRate();                                                     //Translates seconds to samples
      size_t WindowSizeInt = (rate  * mWindowSize);               //Size of window to examine
      size_t SignalWindowSizeInt = (rate  * mSignalWindowSize);   //This much signal is necessary to trip key

      auto samplesleft = len - WindowSizeInt;   //Indexes the number of samples remaining in the selection
      auto lastsubthresholdsample = start;          //start this off at the selection start
      // keeps track of the sample number of the last sample to not exceed the threshold

      int blockruns=0;                         //keeps track of the number of consecutive above-threshold blocks


      //This loop goes through the selection a block at a time.  If a long enough run
      //of above-threshold blocks occur, we return to the last sub-threshold block and
      //go through one sample at a time.
      //If there are fewer than 10 samples leftover, don't bother.

      for(auto i = start; samplesleft >= 10;
          i += (WindowSizeInt - 1) , samplesleft -= (WindowSizeInt - 1)) {

         //Set blocksize so that it is the right size
         const auto blocksize = limitSampleBufferSize( WindowSizeInt, samplesleft);

         //Test whether we are above threshold (the number of stats)
         if(AboveThreshold(t,i,blocksize))
            {
               blockruns++;                   //Hit
            } else {
               blockruns=0;                   //Miss--start over
               lastsubthresholdsample = i;
            }

         //If the blockrun is long enough, break out of the loop early:
         if(blockruns > mSignalWindowSize/mWindowSize)
            break;

      }

      //Now, if we broke out early (samplesleft > 10), go back to the lastsubthresholdsample and look more carefully
      if(samplesleft > 10) {


         //Calculate how many to scan through--we only have to go through (at most)
         //the first window + 1 samples--but we need another window samples to draw from.
         size_t remaining = 2*WindowSizeInt+1;

         //To speed things up, create a local buffer to store things in, to avoid the costly t.Get();
         //Only go through the first SignalWindowSizeInt samples, and choose the first that trips the key.
         float *buffer = new float[remaining];
         t.Get((samplePtr)buffer, floatSample, lastsubthresholdsample, remaining);



         //Initialize these trend markers atrend and ztrend.  They keep track of the
         //up/down trends at the start and end of the evaluation window.
         int atrend = sgn(buffer[1]-buffer[0]);
         int ztrend = sgn(buffer[WindowSizeInt+1]-buffer[WindowSizeInt]);


         double erg=0;
         double  sc=0;
         double  dc=0;

         //Get initial test statistic values.
         if(mUseEnergy)
            erg = TestEnergy(t, lastsubthresholdsample, WindowSizeInt);

         if(mUseSignChangesLow || mUseSignChangesHigh)
            sc  = TestSignChanges(t,lastsubthresholdsample, WindowSizeInt);

         if(mUseDirectionChangesLow || mUseDirectionChangesHigh)
            dc  = TestDirectionChanges(t,lastsubthresholdsample,WindowSizeInt);


         //Now, go through the sound again, sample by sample.
         wxASSERT(WindowSizeInt < SignalWindowSizeInt);
         size_t i;
         for(i = 0; i + WindowSizeInt < SignalWindowSizeInt; i++) {

            int tests = 0;
            int testThreshold = 0;
            //Update the test statistics
            if(mUseEnergy)
               {
//.........这里部分代码省略.........

// 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 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 EffectTruncSilence::Process()
{
   // Typical fraction of total time taken by detection (better to guess low)
   const double detectFrac = .4;

   // Copy tracks
   this->CopyInputTracks(Track::All);

   // Lower bound on the amount of silence to find at a time -- this avoids
   // detecting silence repeatedly in low-frequency sounds.
   const double minTruncMs = 0.001;
   double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];

   // Master list of silent regions; it is responsible for deleting them.
   // This list should always be kept in order.
   RegionList silences;
   silences.DeleteContents(true);

   // Start with the whole selection silent
   Region *sel = new Region;
   sel->start = mT0;
   sel->end = mT1;
   silences.push_back(sel);

   // Remove non-silent regions in each track
   SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
   int whichTrack = 0;
   for (Track *t = iter.First(); t; t = iter.Next())
   {
      WaveTrack *wt = (WaveTrack *)t;

      // Smallest silent region to detect in frames
      sampleCount minSilenceFrames =
            sampleCount(wxMax( mInitialAllowedSilence, minTruncMs) *
                  wt->GetRate());

      //
      // Scan the track for silences
      //
      RegionList trackSilences;
      trackSilences.DeleteContents(true);
      sampleCount blockLen = wt->GetMaxBlockSize();
      sampleCount start = wt->TimeToLongSamples(mT0);
      sampleCount end = wt->TimeToLongSamples(mT1);

      // Allocate buffer
      float *buffer = new float[blockLen];

      sampleCount index = start;
      sampleCount silentFrames = 0;
      bool cancelled = false;

      // Keep position in overall silences list for optimization
      RegionList::iterator rit(silences.begin());

      while (index < end) {
         // Show progress dialog, test for cancellation
         cancelled = TotalProgress(
               detectFrac * (whichTrack + index / (double)end) /
               (double)GetNumWaveTracks());
         if (cancelled)
            break;

         //
         // Optimization: if not in a silent region skip ahead to the next one
         //
         double curTime = wt->LongSamplesToTime(index);
         for ( ; rit != silences.end(); ++rit)
         {
            // Find the first silent region ending after current time
            if ((*rit)->end >= curTime)
               break;
         }

         if (rit == silences.end()) {
            // No more regions -- no need to process the rest of the track
            break;
         }
         else if ((*rit)->start > curTime) {
            // End current silent region, skip ahead
            if (silentFrames >= minSilenceFrames) {
               Region *r = new Region;
               r->start = wt->LongSamplesToTime(index - silentFrames);
               r->end = wt->LongSamplesToTime(index);
               trackSilences.push_back(r);
            }
            silentFrames = 0;

            index = wt->TimeToLongSamples((*rit)->start);
         }
         //
         // End of optimization
         //

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

//.........这里部分代码省略.........