コード例 #1
0
ファイル: ChangeSpeed.cpp プロジェクト: andreipaga/audacity
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;
}
コード例 #2
0
ファイル: NoiseRemoval.cpp プロジェクト: andreipaga/audacity
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;
}
コード例 #3
0
ファイル: Effect.cpp プロジェクト: tuanmasterit/audacity
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);
         }
         gAudioIO->StopStream();

         while (gAudioIO->IsBusy()) {
            ::wxMilliSleep(100);
         }

         delete mProgress;
      }
      else {
         wxMessageBox(_("Error while opening sound device. Please check the output device settings and the project sample rate."),
                      _("Error"), wxOK | wxICON_EXCLAMATION, FocusDialog);
      }
   }

   if (FocusDialog) {
      FocusDialog->SetFocus();
   }

   delete mOutputTracks;
   mOutputTracks = NULL;

   mTracks->Clear(true); // true => delete the tracks
   delete mTracks;

   mTracks = saveTracks;
}
コード例 #4
0
ファイル: Contrast.cpp プロジェクト: MindFy/audacity
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;
}
コード例 #5
0
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++) {
      buffer[tndx] = new float[blockLen];
   }

   // Start processing
   //Track::All is needed because this effect has clear functionality
   this->CopyInputTracks(Track::All); // Set up mOutputTracks.
   SelectedTrackListOfKindIterator iterOut(Track::Wave, mOutputTracks);

   sampleCount index = start;
   sampleCount outTrackOffset = start;
   bool cancelled = false;
   // Reset
   bool ignoringFrames = false;
   bool truncToMinimum = true;  // Ignore the initial samples until we get above the noise floor
   sampleCount consecutiveSilentFrames = 0;
   sampleCount truncIndex = 0;
   sampleCount i = 0;
   sampleCount keep;

   while (index < end) {

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

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

      // Shift over to account for samples remaining from prior block
      sampleCount limit = count+i;

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

         // Is current frame in all tracks below threshold
         bool below = true;
         for (tndx = 0; tndx < tcount; tndx++) {
            if (fabs(buffer[tndx][i]) >= truncDbSilenceThreshold) {
               below = false;
               break;
            }
         }
         // Make sure we cross-fade and output the last silence
         // so we get a smooth transition into whatever follows the selected region
         // Also set the 'truncToMinimum' flag so that the last silence is truncated to the minimum amount
         if(below && ((index+i+1) == end)) {
            below = false;
            truncToMinimum = true;
         }

         // Count frame if it's below threshold
         if (below) {
            consecutiveSilentFrames++;

            // Ignore this frame (equivalent to cutting it)
            // otherwise, keep sample to be part of allowed silence
            if (consecutiveSilentFrames > truncInitialAllowedSilentSamples) {
               ignoringFrames = true;
               continue;
            }
         }
         else {
            if (ignoringFrames == true) {
               // Scale the consectiveSilentFrames so we keep a silence duration
               // which is proportional to the original silence up to the limit
               keep = consecutiveSilentFrames - truncInitialAllowedSilentSamples;
               keep /= mSilenceCompressRatio;

               // The first and last samples always get truncated to the minimum amount
               if(truncToMinimum == true)
                  keep = 0;
               if(keep > truncLongestAllowedSilentSamples)
                  keep = truncLongestAllowedSilentSamples;
               if(keep < 0)
                  keep = 0;

               // Compute the location of the cross-fade to be halfway through the silence
               // with restriction to the samples we still have available to use
               rampInFrames = (truncInitialAllowedSilentSamples - keep + mBlendFrameCount) / 2;
               if(rampInFrames > truncInitialAllowedSilentSamples)
                  rampInFrames = truncInitialAllowedSilentSamples;
               if(rampInFrames < mBlendFrameCount)
                  rampInFrames = mBlendFrameCount;

               // Include the cross-fade samples in the count to make the loop logic easier
               keep += rampInFrames;
               truncIndex -= rampInFrames;

               // back up for cross-fade
               sampleCount curOffset = i - keep;

               if(curOffset < 0) {
                  // This should never happen, but just in case...
                  keep += curOffset - rampInFrames;
                  if(keep < mBlendFrameCount)
                     keep = mBlendFrameCount;
                  curOffset = 0;
               }
               if(truncIndex < 0) {
                  // This should never happen, but just in case...
                  truncIndex = 0;
               }

               for (tndx = 0; tndx < tcount; tndx++) {
                  // Cross fade the cut point
                  for (fr = 0; fr < mBlendFrameCount; fr++) {
                     buffer[tndx][truncIndex+fr] = ((mBlendFrameCount-fr)*buffer[tndx][truncIndex+fr] + fr*buffer[tndx][curOffset + fr]) / mBlendFrameCount;
                  }
                  // Append the 'keep' samples, if any
                  for ( ; fr < keep; fr++) {
                     buffer[tndx][truncIndex+fr] = buffer[tndx][curOffset + fr];
                  }
               }
               truncIndex += keep;
            }
            consecutiveSilentFrames = 0;
            ignoringFrames = false;
            truncToMinimum = false;
         }

         // Can get here either because > dbThreshold
         // or silence duration isn't longer than allowed
         for (tndx = 0; tndx < tcount; tndx++) {
            buffer[tndx][truncIndex] = buffer[tndx][i];
         }
         truncIndex++;
      }

      // Update tracks if any samples were removed, now or before
      if (outTrackOffset + truncIndex != index + limit) {
         // Put updated sample back into output tracks.
         tndx = 0;
         t = (WaveTrack *) iterOut.First();
         while (t) {
            t->Set((samplePtr)buffer[tndx++], floatSample, outTrackOffset, truncIndex);
            t = (WaveTrack *) iterOut.Next();
         }
      }

      // If currently in a silent section, retain samples for the next pass
      if(ignoringFrames) {
         keep = consecutiveSilentFrames - truncInitialAllowedSilentSamples;
         if(keep > (truncLongestAllowedSilentSamples+mBlendFrameCount))
            keep = truncLongestAllowedSilentSamples+mBlendFrameCount;
         for (tndx = 0; tndx < tcount; tndx++) {
            for(fr = 0; fr < truncInitialAllowedSilentSamples; fr++) {
               buffer[tndx][fr] = buffer[tndx][truncIndex-truncInitialAllowedSilentSamples+fr];
            }
            for(fr = 0; fr < keep; fr++) {
               buffer[tndx][truncInitialAllowedSilentSamples+fr] = buffer[tndx][i-keep+fr];
            }
         }
         // Update the output index, less what we are retaining for next time
         outTrackOffset += truncIndex - truncInitialAllowedSilentSamples;
         // Append the following buffer to the existing data
         i = consecutiveSilentFrames = truncInitialAllowedSilentSamples + keep;
         truncIndex = truncInitialAllowedSilentSamples;
      } else {
         // Maintain output index
         outTrackOffset += truncIndex;
         // Reset the buffer pointers to the beginning
         i = 0;
         truncIndex = 0;
         consecutiveSilentFrames = 0;
      }

      // Update progress and bail if user cancelled
      cancelled = TrackProgress(0, ((double)index / (double)end));
      if (cancelled) {
         break;
      }

      // Bump to next block
      index += count;
   }

   AudacityProject *p = GetActiveProject();
   if (!p)
      return false;

   // Remove stale data at end of output tracks.
   if (!cancelled && (outTrackOffset < end)) {
      t = (WaveTrack *) iterOut.First();
      if( p->IsSticky() )
         t->Clear(outTrackOffset / rate, t1, mOutputTracks);
      else
         while(t) {
            t->Clear(outTrackOffset / rate, t1, mOutputTracks);
            t = (WaveTrack *) iterOut.Next();
         }         

      t1 = outTrackOffset / rate;
   }

   // Free buffers
   for (tndx = 0; tndx < tcount; tndx++) {
      delete [] buffer[tndx];
   }
   delete [] buffer;

   mT0 = t0;
   mT1 = t1;

   this->ReplaceProcessedTracks(!cancelled); 
   return !cancelled;
}
コード例 #6
0
ファイル: ChangeSpeed.cpp プロジェクト: ruthmagnus/audacity
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;
}
コード例 #7
0
ファイル: ChangePitch.cpp プロジェクト: Avi2011class/audacity
// 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();
}
コード例 #8
0
ファイル: Repair.cpp プロジェクト: finefin/audacity
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;
}
コード例 #9
0
ファイル: TruncSilence.cpp プロジェクト: ruthmagnus/audacity
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++) {

         // Is current frame in all tracks below threshold
         bool below = true;
         for (tndx = 0; tndx < tcount; tndx++) {
            if (fabs(buffer[tndx][i]) >= truncDbSilenceThreshold) {
               below = false;
               break;
            }
         }

         // Count frame if it's below threshold
         if (below) {
            consecutiveSilentFrames++;

            // Ignore this frame (equivalent to cutting it)
            // otherwise, keep sample to be part of allowed silence
            if (consecutiveSilentFrames > truncLongestAllowedSilentSamples) {
               ignoringFrames = true;
               continue;
            }
         }
         else {
            if (ignoringFrames == true) {
               sampleCount curOffset = i - rampInFrames;
               truncIndex -= rampInFrames; // backup into ignored frames

               for (tndx = 0; tndx < tcount; tndx++) {
                  sampleCount trunci = truncIndex;
                  for (int fr = 0; fr < rampInFrames; fr++) {
                     buffer[tndx][trunci++] = buffer[tndx][curOffset + fr];
                  }
                  if(((trunci - rampInFrames) - mBlendFrameCount) >= 0) {
                     BlendFrames(buffer[tndx], mBlendFrameCount,
                             ((trunci - rampInFrames) - mBlendFrameCount), 
                             ((i - rampInFrames) - mBlendFrameCount));
                  }
               }
               truncIndex += rampInFrames;
            }
            consecutiveSilentFrames = 0;
            ignoringFrames = false;
         }

         // Can get here either because > dbThreshold
         // or silence duration isn't longer than allowed
         for (tndx = 0; tndx < tcount; tndx++) {
            buffer[tndx][truncIndex] = buffer[tndx][i];
         }
         truncIndex++;
      }

      // Update tracks if any samples were removed
      if (truncIndex < limit) {

         // Put updated sample back into output tracks.
         tndx = 0;
         t = (WaveTrack *) iterOut.First();
         while (t) {
            t->Set((samplePtr)buffer[tndx++], floatSample, outTrackOffset, truncIndex);
            t = (WaveTrack *) iterOut.Next();
         }
      }

      // Maintain output index
      outTrackOffset += truncIndex;

      // Update progress and bail if user cancelled
      cancelled = TrackProgress(0, ((double)index / (double)end));
      if (cancelled) {
         break;
      }

      // Bump to next block
      index += limit;
   }

   // Remove stale data at end of output tracks.
   if (!cancelled && (outTrackOffset < end)) {
      t = (WaveTrack *) iterOut.First();
      while (t) {
         t->Clear(outTrackOffset / rate, t1);
         t = (WaveTrack *) iterOut.Next();
      }
      t1 = outTrackOffset / rate;
   }

   // Free buffers
   for (tndx = 0; tndx < tcount; tndx++) {
      delete [] buffer[tndx];
   }
   delete [] buffer;

   mT0 = t0;
   mT1 = t1;

   this->ReplaceProcessedWaveTracks(!cancelled); 
   return !cancelled;
}
コード例 #10
0
ファイル: SBSMSEffect.cpp プロジェクト: QuincyPYoung/audacity
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;
              outResampleCB = resampleCB;
              rb.offset = start;
              rb.end = end;
              rb.iface = new SBSMSInterfaceSliding(&rateSlide,&pitchSlide,
                                                       bPitchReferenceInput,
                                                       samplesToProcess,0,
                                                       NULL);
               
             
            } else {
              rb.bPitch = false;
              outSlideType = (srProcess==srTrack?SlideIdentity:SlideConstant);
              outResampleCB = postResampleCB;
              rb.ratio = srProcess/srTrack;
              rb.quality = new SBSMSQuality(&SBSMSQualityStandard);
              rb.resampler = new Resampler(resampleCB, &rb, srProcess==srTrack?SlideIdentity:SlideConstant);
              rb.sbsms = new SBSMS(rightTrack?2:1,rb.quality,true);
              rb.SBSMSBlockSize = rb.sbsms->getInputFrameSize();
              rb.SBSMSBuf = (audio*)calloc(rb.SBSMSBlockSize,sizeof(audio));

              processPresamples = wxMin(rb.quality->getMaxPresamples(),
                                        (long)((float)(start-trackStart)*(srProcess/srTrack)));
              trackPresamples = wxMin(start-trackStart,
                                      (long)((float)(processPresamples)*(srTrack/srProcess)));
              rb.offset = start - trackPresamples;
              rb.end = trackEnd;
              rb.iface = new SBSMSEffectInterface(rb.resampler,
                                                      &rateSlide,&pitchSlide,
                                                      bPitchReferenceInput,
                                                      samplesToProcess,processPresamples,
                                                      rb.quality);
            }
            
            Resampler resampler(outResampleCB,&rb,outSlideType);

            audio outBuf[SBSMSOutBlockSize];
            float outBufLeft[2*SBSMSOutBlockSize];
            float outBufRight[2*SBSMSOutBlockSize];

            // Samples in output after SBSMS
            sampleCount samplesToOutput = rb.iface->getSamplesToOutput();

            // Samples in output after resampling back
            sampleCount samplesOut = (sampleCount) ((float)samplesToOutput * (srTrack/srProcess));

            // Duration in track time
            double duration =  (mCurT1-mCurT0) * mTotalStretch;

            if(duration > maxDuration)
               maxDuration = duration;

            TimeWarper *warper = createTimeWarper(mCurT0,mCurT1,maxDuration,rateStart,rateEnd,rateSlideType);
            SetTimeWarper(warper);

            rb.outputLeftTrack = mFactory->NewWaveTrack(leftTrack->GetSampleFormat(),
                                                        leftTrack->GetRate());
            if(rightTrack)
               rb.outputRightTrack = mFactory->NewWaveTrack(rightTrack->GetSampleFormat(),
                                                            rightTrack->GetRate());
            long pos = 0;
            long outputCount = -1;

            // process
            while(pos<samplesOut && outputCount) {
               long frames;
               if(pos+SBSMSOutBlockSize>samplesOut) {
                  frames = samplesOut - pos;
               } else {
                  frames = SBSMSOutBlockSize;
               }
               outputCount = resampler.read(outBuf,frames);
               for(int i = 0; i < outputCount; i++) {
                  outBufLeft[i] = outBuf[i][0];
                  if(rightTrack)
                     outBufRight[i] = outBuf[i][1];
               }
               pos += outputCount;
               rb.outputLeftTrack->Append((samplePtr)outBufLeft, floatSample, outputCount);
               if(rightTrack)
                  rb.outputRightTrack->Append((samplePtr)outBufRight, floatSample, outputCount);

               double frac = (double)pos/(double)samplesOut;
               int nWhichTrack = mCurTrackNum;
               if(rightTrack) {
                  nWhichTrack = 2*(mCurTrackNum/2);
                  if (frac < 0.5)
                     frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once.
                  else {
                     nWhichTrack++;
                     frac -= 0.5;
                     frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once.
                  }
               }
               if (TrackProgress(nWhichTrack, frac))
                  return false;
            }
            rb.outputLeftTrack->Flush();
            if(rightTrack)
               rb.outputRightTrack->Flush();

            bool bResult =
               leftTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputLeftTrack,
                                          true, false, GetTimeWarper());
            wxASSERT(bResult); // TO DO: Actually handle this.
            wxUnusedVar(bResult);

            if(rightTrack)
            {
               bResult =
                  rightTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputRightTrack,
                                             true, false, GetTimeWarper());
               wxASSERT(bResult); // TO DO: Actually handle this.
            }
         }
         mCurTrackNum++;
      }
      else if (mustSync && t->IsSyncLockSelected())
      {
         t->SyncLockAdjust(mCurT1, mCurT0 + (mCurT1 - mCurT0) * mTotalStretch);
      }
      //Iterate to the next track
      t = iter.Next();
   }

   if (bGoodResult)
      ReplaceProcessedTracks(bGoodResult);

   // Update selection
   mT0 = mCurT0;
   mT1 = mCurT0 + maxDuration;

   return bGoodResult;
}
コード例 #11
0
ファイル: Repair.cpp プロジェクト: GYGit/Audacity
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;
}
コード例 #12
0
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;
               }
            }
         }
         mCurTrackNum++;
      }
      else if (mustSync && t->IsSyncLockSelected()) {
         t->SyncLockAdjust(mT1, GetTimeWarper()->Warp(mT1));
      }

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

   if (bGoodResult)
      ReplaceProcessedTracks(bGoodResult);

   delete mSoundTouch;
   mSoundTouch = NULL;

//   mT0 = mCurT0;
//   mT1 = mCurT0 + m_maxNewLength; // Update selection.

   return bGoodResult;
}
コード例 #13
0
ファイル: SBSMSEffect.cpp プロジェクト: ruthmagnus/audacity
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
            sampleCount samplesToProcess = (sampleCount) ((real)samplesIn*(srSBSMS/srIn));
            
            // Samples in output after resampling back
            sampleCount samplesToGenerate = (sampleCount) ((real)samplesToProcess * mTotalStretch);
            sampleCount samplesOut = (sampleCount) ((real)samplesIn * mTotalStretch);
            double duration =  (mCurT1-mCurT0) * mTotalStretch;

            if(duration > maxDuration)
               maxDuration = duration;

            TimeWarper *warper = NULL;
            if (rateStart == rateEnd)
            {
               warper = new LinearTimeWarper(mCurT0, mCurT0,
                                             mCurT1, mCurT0+maxDuration);
            } else
            {
               warper = new LogarithmicTimeWarper(mCurT0, mCurT1,
                                                  rateStart, rateEnd);
            }
            SetTimeWarper(warper);
            
            sbsmsInfo si;
            si.rs = rb.resampler;
            si.samplesToProcess = samplesToProcess;
            si.samplesToGenerate = samplesToGenerate;
            si.stretch0 = rateStart;
            si.stretch1 = rateEnd;
            si.ratio0 = pitchStart;
            si.ratio1 = pitchEnd;
            
            rb.sbsmser = sbsms_create(&samplesCB,&stretchCB,&ratioCB,rightTrack?2:1,quality,bPreAnalyze,true);
            rb.pitch = pitch_create(rb.sbsmser,&si,srIn/srSBSMS);
            
            rb.outputLeftTrack = mFactory->NewWaveTrack(leftTrack->GetSampleFormat(),
                                                        leftTrack->GetRate());
            if(rightTrack)
               rb.outputRightTrack = mFactory->NewWaveTrack(rightTrack->GetSampleFormat(),
                                                            rightTrack->GetRate());
            
            
            sampleCount blockSize = SBSMS_FRAME_SIZE[quality];
            rb.outBuf = (audio*)calloc(blockSize,sizeof(audio));
            rb.outputLeftBuffer = (float*)calloc(blockSize*2,sizeof(float));
            if(rightTrack)
               rb.outputRightBuffer = (float*)calloc(blockSize*2,sizeof(float));
            
            long pos = 0;
            long outputCount = -1;
            
            // pre analysis
            real fracPre = 0.0f;
            if(bPreAnalyze) {
               fracPre = 0.05f;
               resampleBuf rbPre;
               rbPre.block = maxBlockSize;
               rbPre.buf = (audio*)calloc(rb.block,sizeof(audio));
               rbPre.leftTrack = leftTrack;
               rbPre.rightTrack = rightTrack?rightTrack:leftTrack;
               rbPre.leftBuffer = (float*)calloc(maxBlockSize,sizeof(float));
               rbPre.rightBuffer = (float*)calloc(maxBlockSize,sizeof(float));
               rbPre.offset = start;
               rbPre.end = end;
               rbPre.ratio = srSBSMS/srIn;
               rbPre.resampler = new Resampler(resampleCB, &rbPre);
               si.rs = rbPre.resampler;
               
               long pos = 0;
               long lastPos = 0;
               long ret = 0;
               while(lastPos<samplesToProcess) {
                  ret = sbsms_pre_analyze(&samplesCB,&si,rb.sbsmser);
                  lastPos = pos;
                  pos += ret;
                  real completion = (real)lastPos/(real)samplesToProcess;
                  if (TrackProgress(0,fracPre*completion))
                     return false;
               }
               sbsms_pre_analyze_complete(rb.sbsmser);
               sbsms_reset(rb.sbsmser);
               si.rs = rb.resampler;
            }
            
            // process
            while(pos<samplesOut && outputCount) {
               long frames;
               if(pos+blockSize>samplesOut) {
                  frames = samplesOut - pos;
               } else {
                  frames = blockSize;
               }
               
               outputCount = pitch_process(rb.outBuf, frames, rb.pitch);
               for(int i = 0; i < outputCount; i++) {
                  rb.outputLeftBuffer[i] = rb.outBuf[i][0];
                  if(rightTrack)
                     rb.outputRightBuffer[i] = rb.outBuf[i][1];
               }
               pos += outputCount;
               rb.outputLeftTrack->Append((samplePtr)rb.outputLeftBuffer, floatSample, outputCount);
               if(rightTrack)
                  rb.outputRightTrack->Append((samplePtr)rb.outputRightBuffer, floatSample, outputCount);
               
               double frac = (double)pos/(double)samplesOut;
               int nWhichTrack = mCurTrackNum;
               if(rightTrack) {
                  nWhichTrack = 2*(mCurTrackNum/2);
                  if (frac < 0.5)
                     frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once. 
                  else {
                     nWhichTrack++;
                     frac -= 0.5;
                     frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once. 
                  }
               }
               if (TrackProgress(nWhichTrack, fracPre + (1.0-fracPre)*frac))
                  return false;
            }
            rb.outputLeftTrack->Flush();
            if(rightTrack)
               rb.outputRightTrack->Flush();
            
            leftTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputLeftTrack,
                  true, false, GetTimeWarper());

            if(rightTrack) {
               rightTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputRightTrack,
                     true, false, GetTimeWarper());
            }
         }
         mCurTrackNum++;
      }
      else if (mustSync && t->IsSynchroSelected())
      {
         t->SyncAdjust(mCurT1, mCurT0 + (mCurT1 - mCurT0) * mTotalStretch);
      }
      //Iterate to the next track
      t = iter.Next();
   }
   
   if (bGoodResult)
      ReplaceProcessedTracks(bGoodResult); 

   // Update selection
   mT0 = mCurT0;
   mT1 = mCurT0 + maxDuration;
   
   return bGoodResult;
}
コード例 #14
0
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;
}
コード例 #15
0
ファイル: Contrast.cpp プロジェクト: disinteger1/audacity
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;
}
コード例 #16
0
ファイル: Normalize.cpp プロジェクト: tuanmasterit/audacity
bool EffectNormalize::Process()
{
   if (mGain == false && mDC == false)
      return true;

   float ratio;
   if( mGain )
      ratio = pow(10.0,TrapDouble(mLevel, // same value used for all tracks
                               NORMALIZE_DB_MIN,
                               NORMALIZE_DB_MAX)/20.0);
   else
      ratio = 1.0;

   //Iterate over each track
   this->CopyInputTracks(); // Set up mOutputTracks.
   bool bGoodResult = true;
   SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks);
   WaveTrack *track = (WaveTrack *) iter.First();
   WaveTrack *prevTrack;
   prevTrack = track;
   mCurTrackNum = 0;
   wxString topMsg;
   if(mDC & mGain)
      topMsg = _("Removing DC offset and Normalizing...\n");
   else if(mDC & !mGain)
      topMsg = _("Removing DC offset...\n");
   else if(!mDC & mGain)
      topMsg = _("Normalizing without removing DC offset...\n");
   else if(!mDC & !mGain)
      topMsg = wxT("Not doing anything)...\n");   // shouldn't get here

   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:
      mCurT0 = mT0 < trackStart? trackStart: mT0;
      mCurT1 = mT1 > trackEnd? trackEnd: mT1;

      // Process only if the right marker is to the right of the left marker
      if (mCurT1 > mCurT0) {
         wxString msg;
         wxString trackName = track->GetName();

         if(!track->GetLinked() || mStereoInd)
            msg = topMsg + _("Analyzing: ") + trackName;
         else
            msg = topMsg + _("Analyzing first track of stereo pair: ") + trackName;
         AnalyseTrack(track, msg);  // sets mOffset and offset-adjusted mMin and mMax
         if(!track->GetLinked() || mStereoInd) {   // mono or 'stereo tracks independently'
            float extent = wxMax(fabs(mMax), fabs(mMin));
            if( (extent > 0) && mGain )
               mMult = ratio / extent;
            else
               mMult = 1.0;
            msg = topMsg + _("Processing: ") + trackName;
            if(track->GetLinked() || prevTrack->GetLinked())  // only get here if there is a linked track but we are processing independently
               msg = topMsg + _("Processing stereo channels independently: ") + trackName;

            if (!ProcessOne(track, msg))
            {
               bGoodResult = false;
               break;
            }
         }
         else
         {
            // we have a linked stereo track
            // so we need to find it's min, max and offset
            // as they are needed to calc the multiplier for both tracks
            float offset1 = mOffset;   // remember ones from first track
            float min1 = mMin;
            float max1 = mMax;
            track = (WaveTrack *) iter.Next();  // get the next one
            mCurTrackNum++;   // keeps progress bar correct
            msg = topMsg + _("Analyzing second track of stereo pair: ") + trackName;
            AnalyseTrack(track, msg);  // sets mOffset and offset-adjusted mMin and mMax
            float offset2 = mOffset;   // ones for second track
            float min2 = mMin;
            float max2 = mMax;
            float extent = wxMax(fabs(min1), fabs(max1));
            extent = wxMax(extent, fabs(min2));
            extent = wxMax(extent, fabs(max2));
            if( (extent > 0) && mGain )
               mMult = ratio / extent; // we need to use this for both linked tracks
            else
               mMult = 1.0;
            mOffset = offset1;
            track = (WaveTrack *) iter.Prev();  // go back to the first linked one
            mCurTrackNum--;   // keeps progress bar correct
            msg = topMsg + _("Processing first track of stereo pair: ") + trackName;
            if (!ProcessOne(track, msg))
            {
               bGoodResult = false;
               break;
            }
            mOffset = offset2;
            track = (WaveTrack *) iter.Next();  // go to the second linked one
            mCurTrackNum++;   // keeps progress bar correct
            msg = topMsg + _("Processing second track of stereo pair: ") + trackName;
            if (!ProcessOne(track, msg))
            {
               bGoodResult = false;
               break;
            }
         }
      }
      
      //Iterate to the next track
      prevTrack = track;
      track = (WaveTrack *) iter.Next();
      mCurTrackNum++;
   }

   this->ReplaceProcessedTracks(bGoodResult); 
   return bGoodResult;
}
コード例 #17
0
ファイル: Normalize.cpp プロジェクト: Rubelislam9950/Audacity
bool EffectNormalize::Process()
{
   bool wasLinked = false; // set when a track has a linked (stereo) track

   if (mGain == false &&
       mDC == false)
      return true;

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

   SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks);
   WaveTrack *track = (WaveTrack *) iter.First();
   mCurTrackNum = 0;
   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:
      mCurT0 = mT0 < trackStart? trackStart: mT0;
      mCurT1 = mT1 > trackEnd? trackEnd: mT1;

      // 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 = track->TimeToLongSamples(mCurT0);
         sampleCount end = track->TimeToLongSamples(mCurT1);
         
         //Get the track rate and samples
         mCurRate = track->GetRate();
         mCurChannel = track->GetChannel();

         if(mStereoInd) // do stereo tracks independently (the easy way)
            track->GetMinMax(&mMin, &mMax, mCurT0, mCurT1);
         else
         {
            if(!wasLinked) // new mono track or first of a stereo pair
            {
               track->GetMinMax(&mMin, &mMax, mCurT0, mCurT1);
               if(track->GetLinked())
               {
                  wasLinked = true; // so we use these values for the next (linked) track
                  track = (WaveTrack *) iter.Next();  // get the next one for the max/min
                  float min, max;
                  track->GetMinMax(&min, &max, mCurT0, mCurT1);
                  mMin = min < mMin ? min : mMin;
                  mMax = max > mMax ? max : mMax;
                  track = (WaveTrack *) iter.Prev();  // back to the one we are on
               }
            }
            else
               wasLinked = false;   // second of the stereo pair, next one is mono or first
         }

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

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