C++ WaveTrack::GetSampleFormat方法代码示例

bool EffectNoise::Process()
{
   if (noiseDuration <= 0.0)
      noiseDuration = sDefaultGenerateLen;

   //Iterate over each track
   TrackListIterator iter(mWaveTracks);
   WaveTrack *track = (WaveTrack *)iter.First();
   while (track) {
      WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());
      numSamples = (longSampleCount)(noiseDuration * track->GetRate() + 0.5);
      longSampleCount i = 0;
      float *data = new float[tmp->GetMaxBlockSize()];
      sampleCount block;

      while(i < numSamples) {
         block = tmp->GetBestBlockSize(i);
         if (block > (numSamples - i))
             block = numSamples - i;

         MakeNoise(data, block, track->GetRate(), noiseAmplitude);

         tmp->Append((samplePtr)data, floatSample, block);
         i += block;
      }
      delete[] data;

      tmp->Flush();
      track->Clear(mT0, mT1);
      track->Paste(mT0, tmp);
      delete tmp;

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

   /*
      save last used values
      save duration unless value was got from selection, so we save only
      when user explicitely setup a value
   */
   if (mT1 == mT0)
      gPrefs->Write(wxT("/CsPresets/NoiseGen_Duration"), noiseDuration);

   gPrefs->Write(wxT("/CsPresets/NoiseGen_Type"), noiseType);
   gPrefs->Write(wxT("/CsPresets/NoiseGen_Amp"), noiseAmplitude);

   mT1 = mT0 + noiseDuration; // Update selection.

   return true;
}

bool EffectToneGen::Process()
{
   if (length <= 0.0)
      length = sDefaultGenerateLen;

   //Iterate over each track
   TrackListIterator iter(mWaveTracks);
   WaveTrack *track = (WaveTrack *)iter.First();
   while (track) {
      mSample = 0;
      WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat());
      mCurRate = track->GetRate();
      tmp->SetRate(mCurRate);
      longSampleCount numSamples =
         (longSampleCount)(length * mCurRate + 0.5);
      longSampleCount i = 0;
      float *data = new float[tmp->GetMaxBlockSize()];
      sampleCount block;

      while(i < numSamples) {
         block = tmp->GetBestBlockSize(i);
         if (block > (numSamples - i))
             block = numSamples - i;
         MakeTone(data, block);
         tmp->Append((samplePtr)data, floatSample, block);
         i += block;
      }
      delete[] data;

      tmp->Flush();
      track->Clear(mT0, mT1);
      track->Paste(mT0, tmp);
      delete tmp;
      
      //Iterate to the next track
      track = (WaveTrack *)iter.Next();
   }

	mT1 = mT0 + length; // Update selection.

   return true;
}

bool EffectSilence::Process()
{
   if (length <= 0.0)
      length = sDefaultGenerateLen;

   TrackListIterator iter(mWaveTracks);
   WaveTrack *track = (WaveTrack *)iter.First();
   while (track) {
      WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());
      tmp->InsertSilence(0.0, length);
      tmp->Flush();
      track->Clear(mT0, mT1);
      track->Paste(mT0, tmp);
      delete tmp;
      
      //Iterate to the next track
      track = (WaveTrack *)iter.Next();
   }

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

bool Generator::Process()
{
   if (GetDuration() < 0.0)
      return false;


   // Set up mOutputTracks.
   // This effect needs Track::All for sync-lock grouping.
   this->CopyInputTracks(Track::All);

   // Iterate over the tracks
   bool bGoodResult = true;
   int ntrack = 0;
   TrackListIterator iter(mOutputTracks);
   Track* t = iter.First();

   while (t != NULL)
   {
      if (t->GetKind() == Track::Wave && t->GetSelected()) {
         WaveTrack* track = (WaveTrack*)t;

         bool editClipCanMove;
         gPrefs->Read(wxT("/GUI/EditClipCanMove"), &editClipCanMove, true);

         //if we can't move clips, and we're generating into an empty space,
         //make sure there's room.
         if (!editClipCanMove &&
             track->IsEmpty(mT0, mT1+1.0/track->GetRate()) &&
             !track->IsEmpty(mT0, mT0+GetDuration()-(mT1-mT0)-1.0/track->GetRate()))
         {
            wxMessageBox(
                  _("There is not enough room available to generate the audio"),
                  _("Error"), wxICON_STOP);
            Failure();
            return false;
         }

         if (GetDuration() > 0.0)
         {
            AudacityProject *p = GetActiveProject();
            // Create a temporary track
            std::unique_ptr<WaveTrack> tmp(
               mFactory->NewWaveTrack(track->GetSampleFormat(),
               track->GetRate())
            );
            BeforeTrack(*track);
            BeforeGenerate();

            // Fill it with data
            if (!GenerateTrack(&*tmp, *track, ntrack))
               bGoodResult = false;
            else {
               // Transfer the data from the temporary track to the actual one
               tmp->Flush();
               SetTimeWarper(new StepTimeWarper(mT0+GetDuration(), GetDuration()-(mT1-mT0)));
               bGoodResult = track->ClearAndPaste(p->GetSel0(), p->GetSel1(), &*tmp, true,
                     false, GetTimeWarper());
            }

            if (!bGoodResult) {
               Failure();
               return false;
            }
         }
         else
         {
            // If the duration is zero, there's no need to actually
            // generate anything
            track->Clear(mT0, mT1);
         }

         ntrack++;
      }
      else if (t->IsSyncLockSelected()) {
         t->SyncLockAdjust(mT1, mT0 + GetDuration());
      }
      // Move on to the next track
      t = iter.Next();
   }

   Success();

   this->ReplaceProcessedTracks(bGoodResult);

   mT1 = mT0 + GetDuration(); // Update selection.

   return true;
}

//------------------------- Processing methods -------------------------
bool EffectSineSweepGenerator::Process()
{
   // taken `as is` from Audacity`s Generator.cpp to resolve
   // a vc++ linking problem...
   
   if (mDuration < 0.0)
      return false;

   BeforeGenerate();

   // Set up mOutputTracks. This effect needs Track::All for grouping
   this->CopyInputTracks(Track::All);

   // Iterate over the tracks
   bool bGoodResult = true;
   int ntrack = 0;
   TrackListIterator iter(mOutputTracks);
   Track* t = iter.First();

   while (t != NULL)
   {
      if (t->GetKind() == Track::Wave && t->GetSelected()) 
      {
         WaveTrack* track = (WaveTrack*)t;
         
         bool editClipCanMove = true;
         //gPrefs->Read(wxT("/GUI/EditClipCanMove"), &editClipCanMove, true);

         //if we can't move clips, and we're generating into an empty space,
         //make sure there's room.
         if (!editClipCanMove &&
             track->IsEmpty(mT0, mT1+1.0/track->GetRate()) &&
             !track->IsEmpty(mT0, mT0+mDuration-(mT1-mT0)-1.0/track->GetRate()))
         {
             wxMessageBox(_("There is not enough room available to generate the audio"),
                          _("Error"), wxICON_STOP);   
            Failure();
            return false;
         }

         if (mDuration > 0.0)
         {
            // Create a temporary track
            WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(),
                                                    track->GetRate());
            //BeforeTrack(*track);

            // Fill it with data
            if (!GenerateTrack(tmp, *track, ntrack))
               bGoodResult = false;
            else 
            {
               // Transfer the data from the temporary track to the actual one
               tmp->Flush();
               SetTimeWarper(new AFStepTimeWarper(mT0+mDuration, mDuration-(mT1-mT0)));
               bGoodResult = track->ClearAndPaste(mT0, mT1, tmp, true,
                     false, GetTimeWarper());
               delete tmp;
            }

            if (!bGoodResult) 
            {
               Failure();
               return false;
            }
         }
         else
         {
            // If the duration is zero, there's no need to actually
            // generate anything
            track->Clear(mT0, mT1);
         }

         ntrack++;
      }
      else if (t->IsSyncLockSelected()) 
      {
         t->SyncLockAdjust(mT1, mT0 + mDuration);
      }
      // Move on to the next track
      t = iter.Next();
   }

   Success();

   this->ReplaceProcessedTracks(bGoodResult);

   mT1 = mT0 + mDuration; // Update selection.

   return true;
}

bool EffectNoise::Process()
{
   if (noiseDuration <= 0.0)
      noiseDuration = sDefaultGenerateLen;

   //Iterate over each track
   int ntrack = 0;
   this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
   bool bGoodResult = true;

   TrackListIterator iter(mOutputWaveTracks);
   WaveTrack *track = (WaveTrack *)iter.First();
   while (track) {
      WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());
      numSamples = (longSampleCount)(noiseDuration * track->GetRate() + 0.5);
      longSampleCount i = 0;
      float *data = new float[tmp->GetMaxBlockSize()];
      sampleCount block;

      while ((i < numSamples) && bGoodResult) {
         block = tmp->GetBestBlockSize(i);
         if (block > (numSamples - i))
             block = numSamples - i;

         MakeNoise(data, block, track->GetRate(), noiseAmplitude);

         tmp->Append((samplePtr)data, floatSample, block);
         i += block;

         //Update the Progress meter
         if (TrackProgress(ntrack, (double)i / numSamples))
            bGoodResult = false;
      }
      delete[] data;

      tmp->Flush();
      track->Clear(mT0, mT1);
      track->Paste(mT0, tmp);
      delete tmp;

      if (!bGoodResult)
         break;

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

   if (bGoodResult)
   {
      /*
         save last used values
         save duration unless value was got from selection, so we save only
         when user explicitely setup a value
      */
      if (mT1 == mT0)
         gPrefs->Write(wxT("/CsPresets/NoiseGen_Duration"), noiseDuration);

      gPrefs->Write(wxT("/CsPresets/NoiseGen_Type"), noiseType);
      gPrefs->Write(wxT("/CsPresets/NoiseGen_Amp"), noiseAmplitude);

      mT1 = mT0 + noiseDuration; // Update selection.
   }

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

//.........这里部分代码省略.........
                                        (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.

//.........这里部分代码省略.........
            
            // 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;
               

bool EffectDtmf::Process()
{
   if (dtmfDuration <= 0.0)
      return false;

   //Iterate over each track
   TrackListIterator iter(mWaveTracks);
   WaveTrack *track = (WaveTrack *)iter.First();
   while (track) {
      // new tmp track, to fill with dtmf sequence
      // we will build the track by adding a tone, then a silence, next tone, and so on...
      WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());

      // all dtmf sequence durations in samples from seconds
      numSamplesSequence = (longSampleCount)(dtmfDuration * track->GetRate() + 0.5);
      numSamplesTone = (longSampleCount)(dtmfTone * track->GetRate() + 0.5);
      numSamplesSilence = (longSampleCount)(dtmfSilence * track->GetRate() + 0.5);

      // recalculate the sum, and spread the difference - due to approximations.
      // Since diff should be in the order of "some" samples, a division (resulting in zero)
      // is not sufficient, so we add the additional remaining samples in each tone/silence block,
      // at least until available.
      int diff = numSamplesSequence - (dtmfNTones*numSamplesTone) - (dtmfNTones-1)*numSamplesSilence;
      if (diff>dtmfNTones) {
         // in this case, both these values would change, so it makes sense to recalculate diff
         // otherwise just keep the value we already have

         // should always be the case that dtmfNTones>1, as if 0, we don't even start processing,
         // and with 1 there is no difference to spread (no silence slot)...
         wxASSERT(dtmfNTones > 1);
         numSamplesTone += (diff/(dtmfNTones));
         numSamplesSilence += (diff/(dtmfNTones-1));
         diff = numSamplesSequence - (dtmfNTones*numSamplesTone) - (dtmfNTones-1)*numSamplesSilence;
      }
      // this var will be used as extra samples distributor
      int extra=0;

      longSampleCount i = 0;
      longSampleCount j = 0;
      int n=0; // pointer to string in dtmfString
      sampleCount block;
      bool isTone = true;
      float *data = new float[tmp->GetMaxBlockSize()];

      // for the whole dtmf sequence, we will be generating either tone or silence
      // according to a bool value, and this might be done in small chunks of size
      // 'block', as a single tone might sometimes be larger than the block
      // tone and silence generally have different duration, thus two generation blocks
      //
      // Note: to overcome a 'clicking' noise introduced by the abrupt transition from/to
      // silence, I added a fade in/out of 1/250th of a second (4ms). This can still be
      // tweaked but gives excellent results at 44.1kHz: I haven't tried other freqs.
      // A problem might be if the tone duration is very short (<10ms)... (?)
      //
      // One more problem is to deal with the approximations done when calculating the duration 
      // of both tone and silence: in some cases the final sum might not be same as the initial
      // duration. So, to overcome this, we had a redistribution block up, and now we will spread
      // the remaining samples in every bin in order to achieve the full duration: test case was
      // to generate an 11 tone DTMF sequence, in 4 seconds, and with DutyCycle=75%: after generation
      // you ended up with 3.999s or in other units: 3 seconds and 44097 samples.
      //
      while(i < numSamplesSequence) {
         if (isTone)
         // generate tone
         {
            // the statement takes care of extracting one sample from the diff bin and
            // adding it into the tone block until depletion
            extra=(diff-- > 0?1:0);
            for(j=0; j < numSamplesTone+extra; j+=block) {
               block = tmp->GetBestBlockSize(j);
               if (block > (numSamplesTone+extra - j))
                   block = numSamplesTone+extra - j;

               // generate the tone and append
               MakeDtmfTone(data, block, track->GetRate(), dtmfString[n], j, numSamplesTone);
               tmp->Append((samplePtr)data, floatSample, block);
            }
            i += numSamplesTone;
            n++;
            if(n>=dtmfNTones)break;
         }
         else
         // generate silence
         {
            // the statement takes care of extracting one sample from the diff bin and
            // adding it into the silence block until depletion
            extra=(diff-- > 0?1:0);
            for(j=0; j < numSamplesSilence+extra; j+=block) {
               block = tmp->GetBestBlockSize(j);
               if (block > (numSamplesSilence+extra - j))
                   block = numSamplesSilence+extra - j;

               // generate silence and append
               memset(data, 0, sizeof(float)*block);
               tmp->Append((samplePtr)data, floatSample, block);
            }
            i += numSamplesSilence;
         }
         // flip flag
         isTone=!isTone;
//.........这里部分代码省略.........