コード例 #1
0
ファイル: SoundTouchEffect.cpp プロジェクト: GYGit/Audacity
bool EffectSoundTouch::ProcessStereo(WaveTrack* leftTrack, WaveTrack* rightTrack,
                                     sampleCount start, sampleCount end)
{
   mSoundTouch->setSampleRate((unsigned int)(leftTrack->GetRate()+0.5));

   WaveTrack* outputLeftTrack = mFactory->NewWaveTrack(leftTrack->GetSampleFormat(),
                                                       leftTrack->GetRate());
   WaveTrack* outputRightTrack = mFactory->NewWaveTrack(rightTrack->GetSampleFormat(),
                                                        rightTrack->GetRate());

   //Get the length of the buffer (as double). len is
   //used simple to calculate a progress meter, so it is easier
   //to make it a double now than it is to do it later
   double len = (double)(end - start);

   //Initiate a processing buffer.  This buffer will (most likely)
   //be shorter than the length of the track being processed.
   // Make soundTouchBuffer twice as big as MaxBlockSize for each channel,
   // because Soundtouch wants them interleaved, i.e., each
   // Soundtouch sample is left-right pair.
   sampleCount maxBlockSize = leftTrack->GetMaxBlockSize();
   float* leftBuffer = new float[maxBlockSize];
   float* rightBuffer = new float[maxBlockSize];
   float* soundTouchBuffer = new float[maxBlockSize * 2];

   // Go through the track one stereo buffer at a time.
   // sourceSampleCount counts the sample at which the current buffer starts,
   // per channel.
   sampleCount sourceSampleCount = start;
   while (sourceSampleCount < end) {
      //Get a block of samples (smaller than the size of the buffer)
      sampleCount blockSize = leftTrack->GetBestBlockSize(sourceSampleCount);

      //Adjust the block size if it is the final block in the track
      if (sourceSampleCount + blockSize > end)
         blockSize = end - sourceSampleCount;

      // Get the samples from the tracks and put them in the buffers.
      leftTrack->Get((samplePtr)(leftBuffer), floatSample, sourceSampleCount, blockSize);
      rightTrack->Get((samplePtr)(rightBuffer), floatSample, sourceSampleCount, blockSize);

      // Interleave into soundTouchBuffer.
      for (int index = 0; index < blockSize; index++) {
         soundTouchBuffer[index*2]       = leftBuffer[index];
         soundTouchBuffer[(index*2)+1]   = rightBuffer[index];
      }

      //Add samples to SoundTouch
      mSoundTouch->putSamples(soundTouchBuffer, blockSize);

      //Get back samples from SoundTouch
      unsigned int outputCount = mSoundTouch->numSamples();
      if (outputCount > 0)
         this->ProcessStereoResults(outputCount, outputLeftTrack, outputRightTrack);

      //Increment sourceSampleCount one blockfull of samples
      sourceSampleCount += blockSize;

      //Update the Progress meter
      // mCurTrackNum is left track. Include right track.
      int nWhichTrack = mCurTrackNum;
      double frac = (sourceSampleCount - start) / len;
      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;
   }

   // Tell SoundTouch to finish processing any remaining samples
   mSoundTouch->flush();

   unsigned int outputCount = mSoundTouch->numSamples();
   if (outputCount > 0)
      this->ProcessStereoResults(outputCount, outputLeftTrack, outputRightTrack);

   // Flush the output WaveTracks (since they're buffered, too)
   outputLeftTrack->Flush();
   outputRightTrack->Flush();

   // Clean up the buffers.
   delete [] leftBuffer;
   delete [] rightBuffer;
   delete [] soundTouchBuffer;

   // Take the output tracks and insert in place of the original
   // sample data.
   leftTrack->ClearAndPaste(mCurT0, mCurT1, outputLeftTrack, true, false, GetTimeWarper());
   rightTrack->ClearAndPaste(mCurT0, mCurT1, outputRightTrack, true, false, GetTimeWarper());

   // Track the longest result length
   double newLength = outputLeftTrack->GetEndTime();
   m_maxNewLength = wxMax(m_maxNewLength, newLength);
   newLength = outputRightTrack->GetEndTime();
   m_maxNewLength = wxMax(m_maxNewLength, newLength);

   // Delete the outputTracks now that their data are inserted in place.
   delete outputLeftTrack;
   delete outputRightTrack;

   //Return true because the effect processing succeeded.
   return true;
}
コード例 #2
0
ファイル: SoundTouchEffect.cpp プロジェクト: GYGit/Audacity
//ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
//and executes ProcessSoundTouch on these blocks
bool EffectSoundTouch::ProcessOne(WaveTrack *track,
                                  sampleCount start, sampleCount end)
{
   WaveTrack *outputTrack;
   sampleCount s;

   mSoundTouch->setSampleRate((unsigned int)(track->GetRate()+0.5));

   outputTrack = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());

   //Get the length of the buffer (as double). len is
   //used simple to calculate a progress meter, so it is easier
   //to make it a double now than it is to do it later
   double len = (double)(end - start);

   //Initiate a processing buffer.  This buffer will (most likely)
   //be shorter than the length of the track being processed.
   float *buffer = new float[track->GetMaxBlockSize()];

   //Go through the track one buffer at a time. s counts which
   //sample the current buffer starts at.
   s = start;
   while (s < end) {
      //Get a block of samples (smaller than the size of the buffer)
      sampleCount block = track->GetBestBlockSize(s);

      //Adjust the block size if it is the final block in the track
      if (s + block > end)
         block = end - s;

      //Get the samples from the track and put them in the buffer
      track->Get((samplePtr) buffer, floatSample, s, block);

      //Add samples to SoundTouch
      mSoundTouch->putSamples(buffer, block);

      //Get back samples from SoundTouch
      unsigned int outputCount = mSoundTouch->numSamples();
      if (outputCount > 0) {
         float *buffer2 = new float[outputCount];
         mSoundTouch->receiveSamples(buffer2, outputCount);
         outputTrack->Append((samplePtr)buffer2, floatSample, outputCount);
         delete[] buffer2;
      }

      //Increment s one blockfull of samples
      s += block;

      //Update the Progress meter
      if (TrackProgress(mCurTrackNum, (s - start) / len))
         return false;
   }

   // Tell SoundTouch to finish processing any remaining samples
   mSoundTouch->flush();   // this should only be used for changeTempo - it dumps data otherwise with pRateTransposer->clear();

   unsigned int outputCount = mSoundTouch->numSamples();
   if (outputCount > 0) {
      float *buffer2 = new float[outputCount];
      mSoundTouch->receiveSamples(buffer2, outputCount);
      outputTrack->Append((samplePtr)buffer2, floatSample, outputCount);
      delete[] buffer2;
   }

   // Flush the output WaveTrack (since it's buffered, too)
   outputTrack->Flush();

   // Clean up the buffer
   delete[]buffer;

   // Take the output track and insert it in place of the original
   // sample data
   track->ClearAndPaste(mCurT0, mCurT1, outputTrack, true, false, GetTimeWarper());

   double newLength = outputTrack->GetEndTime();
   m_maxNewLength = wxMax(m_maxNewLength, newLength);

   // Delete the outputTrack now that its data is inserted in place
   delete outputTrack;

   //Return true because the effect processing succeeded.
   return true;
}
コード例 #3
0
ファイル: ChangeSpeed.cpp プロジェクト: ruthmagnus/audacity
// ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
// and calls libsamplerate code on these blocks.
bool EffectChangeSpeed::ProcessOne(WaveTrack * track,
												longSampleCount start, longSampleCount end)
{
	if (track == NULL)
		return false;

	// initialization, per examples of Mixer::Mixer and
   // EffectSoundTouch::ProcessOne

   WaveTrack * outputTrack = mFactory->NewWaveTrack(track->GetSampleFormat(),
                                                    track->GetRate());

   //Get the length of the selection (as double). len is
   //used simple to calculate a progress meter, so it is easier
   //to make it a double now than it is to do it later 
   double len = (double)(end - start);

   // Initiate processing buffers, most likely shorter than 
	//	the length of the selection being processed.
	sampleCount inBufferSize = track->GetMaxBlockSize();

   float * inBuffer = new float[inBufferSize];

   double factor = 100.0 / (100.0 + m_PercentChange);
	sampleCount outBufferSize = 
      (sampleCount)((factor * inBufferSize) + 10);
   float * outBuffer = new float[outBufferSize]; 

	// Set up the resampling stuff for this track.
   Resample resample(true, factor, factor);

   //Go through the track one buffer at a time. samplePos counts which
   //sample the current buffer starts at.
	bool bLoopSuccess = true;
   sampleCount blockSize;
  	longSampleCount samplePos = start;
   while (samplePos < end) {
      //Get a blockSize of samples (smaller than the size of the buffer)
      blockSize = track->GetBestBlockSize(samplePos);

      //Adjust the block size if it is the final block in the track
      if (samplePos + blockSize > end)
         blockSize = end - samplePos;

      //Get the samples from the track and put them in the buffer
      track->Get((samplePtr) inBuffer, floatSample, samplePos, blockSize);

      int inUsed;
      int outgen = resample.Process(factor,
                                    inBuffer,
                                    blockSize,
                                    ((samplePos + blockSize) >= end),
                                    &inUsed,
                                    outBuffer,
                                    outBufferSize);
      if (outgen < 0) {
			bLoopSuccess = false;
			break;
		}

      if (outgen > 0)
         outputTrack->Append((samplePtr)outBuffer, floatSample, 
                             outgen);

      // Increment samplePos
      samplePos += inUsed;

      // Update the Progress meter
      if (TrackProgress(mCurTrackNum, (samplePos - start) / len)) {
			bLoopSuccess = false;
			break;
		}
   }

	// Flush the output WaveTrack (since it's buffered, too)
	outputTrack->Flush();

   // Clean up the buffers
   delete [] inBuffer;
   delete [] outBuffer;

   // Take the output track and insert it in place of the original
   // sample data
	if (bLoopSuccess) {
		track->Clear(mCurT0, mCurT1);
		track->Paste(mCurT0, outputTrack);
	}

	double newLength = outputTrack->GetEndTime(); 
	if (newLength > m_maxNewLength) 
		m_maxNewLength = newLength; 

   // Delete the outputTrack now that its data is inserted in place
   delete outputTrack;

   return bLoopSuccess;
}
コード例 #4
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;
}
コード例 #5
0
void ControlToolBar::OnRecord(wxCommandEvent &evt)
{
   if (gAudioIO->IsBusy()) {
      mRecord->PopUp();
      return;
   }
   AudacityProject *p = GetActiveProject();
   if (p && p->GetCleanSpeechMode()) {
      size_t numProjects = gAudacityProjects.Count();
      bool tracks = (p && !p->GetTracks()->IsEmpty());
      if (tracks || (numProjects > 1)) {
         wxMessageBox(_("Recording in CleanSpeech mode is not possible when a track, or more than one project, is already open."),
            _("Recording not permitted"),
            wxOK | wxICON_INFORMATION,
            this);
         mRecord->PopUp();
         mRecord->Disable();
         return;
      }
   }

   if( evt.GetInt() == 1 ) // used when called by keyboard shortcut. Default (0) ignored.
      mRecord->SetShift(true);
   if( evt.GetInt() == 2 )
      mRecord->SetShift(false);

   SetRecord(true);

   if (p) {
      TrackList *t = p->GetTracks();
      TrackListIterator it(t);
      if(it.First() == NULL)
         mRecord->SetShift(false);
      double t0 = p->GetSel0();
      double t1 = p->GetSel1();
      if (t1 == t0)
         t1 = 1000000000.0;     // record for a long, long time (tens of years)

      /* TODO: set up stereo tracks if that is how the user has set up
       * their preferences, and choose sample format based on prefs */
      WaveTrackArray newRecordingTracks, playbackTracks;
/* REQUIRES PORTMIDI */
//      NoteTrackArray midiTracks;

      bool duplex;
      gPrefs->Read(wxT("/AudioIO/Duplex"), &duplex, true);
            
      if(duplex){
         playbackTracks = t->GetWaveTrackArray(false);
/* REQUIRES PORTMIDI */
//		 midiTracks = t->GetNoteTrackArray(false);
     }
      else {
         playbackTracks = WaveTrackArray();
/* REQUIRES PORTMIDI */
//		 midiTracks = NoteTrackArray();
     }
      
      // If SHIFT key was down, the user wants append to tracks
      int recordingChannels = 0;
      bool shifted = mRecord->WasShiftDown();
      if (shifted) {
         TrackListIterator it(t);
         WaveTrack *wt;
         bool sel = false;
         double allt0 = t0;

         // Find the maximum end time of selected and all wave tracks
         for (Track *tt = it.First(); tt; tt = it.Next()) {
            if (tt->GetKind() == Track::Wave) {
               wt = (WaveTrack *)tt;
               if (wt->GetEndTime() > allt0) {
                  allt0 = wt->GetEndTime();
               }
            
               if (tt->GetSelected()) {
                  sel = true;
                  if (duplex)
                     playbackTracks.Remove(wt);
                  if (wt->GetEndTime() > t0) {
                     t0 = wt->GetEndTime();
                  }
               }
            }
         }

         // Use end time of all wave tracks if none selected
         if (!sel) {
            t0 = allt0;
         }

         // Pad selected/all wave tracks to make them all the same length
         for (Track *tt = it.First(); tt; tt = it.Next()) {
            if (tt->GetKind() == Track::Wave && (tt->GetSelected() || !sel)) {
               wt = (WaveTrack *)tt;
               t1 = wt->GetEndTime();
               if (t1 < t0) {
                  WaveTrack *newTrack = p->GetTrackFactory()->NewWaveTrack();
                  newTrack->InsertSilence(0.0, t0 - t1);
                  newTrack->Flush();
                  wt->Clear(t1, t0);
                  wt->Paste(t1, newTrack);
                  delete newTrack;
               }
               newRecordingTracks.Add(wt);
            }
         }

         t1 = 1000000000.0;     // record for a long, long time (tens of years)
      }
      else {
         recordingChannels = gPrefs->Read(wxT("/AudioIO/RecordChannels"), 2);
         for (int c = 0; c < recordingChannels; c++) {
            WaveTrack *newTrack = p->GetTrackFactory()->NewWaveTrack();

            int initialheight = newTrack->GetHeight();

            newTrack->SetOffset(t0);

            if (recordingChannels <= 2) {
               newTrack->SetHeight(initialheight/recordingChannels);
            }
            else {
               newTrack->SetMinimized(true);
            }

            if (recordingChannels == 2) {
               if (c == 0) {
                  newTrack->SetChannel(Track::LeftChannel);
                  newTrack->SetLinked(true);
               }
               else {
                  newTrack->SetChannel(Track::RightChannel);
                  newTrack->SetTeamed(true);
               }
            }
            else {
               newTrack->SetChannel( Track::MonoChannel );
            }

            newRecordingTracks.Add(newTrack);
         }
         
         // msmeyer: StartStream calls a callback which triggers auto-save, so
         // we add the tracks where recording is done into now. We remove them
         // later if starting the stream fails
         for (unsigned int i = 0; i < newRecordingTracks.GetCount(); i++)
            t->Add(newRecordingTracks[i]);
      }

      int token = gAudioIO->StartStream(playbackTracks,
                                        newRecordingTracks,
/* REQUIRES PORTMIDI */
//                                        midiTracks,
                                        t->GetTimeTrack(),
                                        p->GetRate(), t0, t1, p);

      bool success = (token != 0);
      
      if (success) {
         p->SetAudioIOToken(token);
         mBusyProject = p;
         SetVUMeters(p);
      }
      else {
         // msmeyer: Delete recently added tracks if opening stream fails
         if (!shifted) {
            for (unsigned int i = 0; i < newRecordingTracks.GetCount(); i++) {
               t->Remove(newRecordingTracks[i]);
               delete newRecordingTracks[i];
            }
         }

         // msmeyer: Show error message if stream could not be opened
         wxMessageBox(_("Error while opening sound device. "
            wxT("Please check the input device settings and the project sample rate.")),
                      _("Error"), wxOK | wxICON_EXCLAMATION, this);

         SetPlay(false);
         SetStop(false);
         SetRecord(false);
      }
   }
}