コード例 #1
0
	QScriptValue ProgressDialog::constructor(QScriptContext *context, QScriptEngine *engine)
	{
		ProgressDialog *progressDialog = new ProgressDialog;
		progressDialog->setupConstructorParameters(context, engine, context->argument(0));

		QScriptValueIterator it(context->argument(0));

		while(it.hasNext())
		{
			it.next();

			if(it.name() == "value")
				progressDialog->mProgressDialog->setValue(it.value().toInt32());
			else if(it.name() == "labelText")
				progressDialog->mProgressDialog->setLabelText(it.value().toString());
			else if(it.name() == "minimum")
				progressDialog->mProgressDialog->setMinimum(it.value().toInt32());
			else if(it.name() == "maximum")
				progressDialog->mProgressDialog->setMaximum(it.value().toInt32());
			else if(it.name() == "range")
			{
				progressDialog->mProgressDialog->setMinimum(it.value().property("minimum").toInt32());
				progressDialog->mProgressDialog->setMaximum(it.value().property("maximum").toInt32());
			}
			else if(it.name() == "onCanceled")
				progressDialog->mOnCanceled = it.value();
		}

		return CodeClass::constructor(progressDialog, context, engine);
	}
コード例 #2
0
void VersionSelectDialog::loadList()
{
	ProgressDialog *taskDlg = new ProgressDialog(this);
	Task *loadTask = m_vlist->getLoadTask();
	loadTask->setParent(taskDlg);
	taskDlg->exec(loadTask);
}
コード例 #3
0
ファイル: MainFrm.cpp プロジェクト: ali-howie/zpack
void CMainFrame::OnFileDefrag()
{
    CzpEditorDoc* document = (CzpEditorDoc*)GetActiveDocument();
    ZpExplorer& explorer = document->GetZpExplorer();
    if (!explorer.isOpen())
    {
        return;
    }
    //zp::u64 fragSize = explorer.getPack()->countFragmentSize();
    //if (fragSize == 0)
    //{
    //	::MessageBox(NULL, _T("No fragment found in package."), _T("Note"), MB_OK | MB_ICONINFORMATION);
    //	return;
    //}
    //StringStream tip;
    //tip << _T("You can save ") << fragSize << _T(" bytes, continue?");
    if (::MessageBox(NULL, _T("It will take minutes for large package, continue?"), _T("Note"), MB_YESNO | MB_ICONQUESTION) != IDYES)
    {
        return;
    }

    ProgressDialog progressDlg;
    progressDlg.m_explorer = &(document->GetZpExplorer());
    progressDlg.m_running = true;
    progressDlg.m_params = NULL;
    progressDlg.m_operation = ProgressDialog::OP_DEFRAG;
    progressDlg.m_totalFileSize = explorer.countNodeFileSize(explorer.rootNode());
    progressDlg.DoModal();
}
コード例 #4
0
ファイル: MainWindow.cpp プロジェクト: HackLinux/opencine
void MainWindow::on_pushButton_clicked()
{
    ProgressDialog* progressDialog = new ProgressDialog(nullptr, new DriveTransfer(nullptr, ".", "./Test"));
    progressDialog->show();
    //Console* console = new Console(this);
    //console->show();
}
コード例 #5
0
BOOL CALLBACK ProgressDialog::dlgProc(HWND hWnd, UINT message, WPARAM wParam, LPARAM lParam)
{
	switch(message)
	{
		case WM_INITDIALOG:
		{
			::SetWindowLongPtr(hWnd, GWL_USERDATA, lParam);
			ProgressDialog* dlg = reinterpret_cast<ProgressDialog*>(lParam);
			return dlg->runDlgProc(hWnd, message, wParam, lParam);
		}

        case WM_COMMAND:
            {
                if (IDCANCEL == wParam) {
                    int mbResult = MessageBox(hWnd, _T("Are you sure you wish to abort the current installation/removal?"), _T("Cancel installation / removal?"), MB_YESNO | MB_ICONQUESTION);
                    if (IDYES == mbResult) {
			            ProgressDialog* dlg = reinterpret_cast<ProgressDialog*>(::GetWindowLongPtr(hWnd, GWL_USERDATA));
                        dlg->_cancelToken.triggerCancel();
                    }
                }
                return FALSE;
            }
		default:
		{
			ProgressDialog* dlg = reinterpret_cast<ProgressDialog*>(::GetWindowLongPtr(hWnd, GWL_USERDATA));
			return dlg->runDlgProc(hWnd, message, wParam, lParam);
		}
	}

}
コード例 #6
0
ファイル: zpEditorView.cpp プロジェクト: ali-howie/zpack
void CzpEditorView::startOperation(ProgressDialog::Operation op, zp::u64 totalFileSize,
							const std::vector<std::pair<zp::String, zp::String>>* params)
{
	ProgressDialog progressDlg;
	progressDlg.m_explorer = &(GetDocument()->GetZpExplorer());
	progressDlg.m_running = true;
	progressDlg.m_params = params;
	progressDlg.m_operation = op;
	progressDlg.m_totalFileSize = totalFileSize;
	progressDlg.DoModal();
}
コード例 #7
0
ファイル: localview.cpp プロジェクト: caidongyun/nullfxp
void LocalView::slot_local_new_download_requested(const TaskPackage &local_pkg, const TaskPackage &remote_pkg)
{
    ProgressDialog *pdlg = new ProgressDialog(0);
    // src is remote file , dest if localfile 
    pdlg->set_transfer_info(remote_pkg, local_pkg);
    QObject::connect(pdlg, SIGNAL(transfer_finished(int, QString)),
                     this, SLOT(slot_transfer_finished(int, QString)));
    this->main_mdi_area->addSubWindow(pdlg);
    pdlg->show();
    this->own_progress_dialog = pdlg;
}
コード例 #8
0
// ----------------------------------------------------------------------------
// dialogProc
// ----------------------------------------------------------------------------
// the message procedure for the cgp dialog procedure, a dummy procedure
INT_PTR CALLBACK ProgressDialog::progDialogProc(HWND hWndDlg, UINT msg,
									   WPARAM wParam, LPARAM lParam) {
   //this->hWnd = hWnd;
	if (msg == WM_INITDIALOG) {
		ProgressDialog* progD = reinterpret_cast<ProgressDialog*>(lParam);
		SetWindowLong(hWndDlg, GWL_USERDATA, reinterpret_cast<long>(progD));
		
	}
	// get the WWindow structure and pass the data to that
	ProgressDialog* progD = (ProgressDialog *) GetWindowLong(hWndDlg, GWL_USERDATA); 
	if (progD)
		return progD->realProgDialogProc(hWndDlg, msg, wParam, lParam);
	
										   
	
} // --- dialogProc -----------------------------------------------------------
コード例 #9
0
BOOL CALLBACK ProgressDialog::dlgProc(HWND hWnd, UINT message, WPARAM wParam, LPARAM lParam)
{
	switch(message)
	{
		case WM_INITDIALOG:
		{
			::SetWindowLongPtr(hWnd, GWLP_USERDATA, lParam);
			ProgressDialog* dlg = reinterpret_cast<ProgressDialog*>(lParam);
			return dlg->runDlgProc(hWnd, message, wParam, lParam);
		}
		default:
		{
			ProgressDialog* dlg = reinterpret_cast<ProgressDialog*>(static_cast<LONG_PTR>(::GetWindowLongPtr(hWnd, GWLP_USERDATA)));
			return dlg->runDlgProc(hWnd, message, wParam, lParam);
		}
	}

}
コード例 #10
0
ファイル: localview.cpp プロジェクト: caidongyun/nullfxp
void LocalView::slot_transfer_finished(int status, QString errorString)
{
    qDebug() <<__FUNCTION__<<": "<<__LINE__<<":"<< __FILE__<<status; 
    // SFTPView *remote_view = this;
    
    ProgressDialog *pdlg = (ProgressDialog*)sender();

    this->main_mdi_area->removeSubWindow(pdlg->parentWidget());

    delete pdlg;
    this->own_progress_dialog = NULL;

    if (status == 0 // || status == 3
        ) {
        //TODO 通知UI更新目录结构,在某些情况下会导致左侧树目录变空。
        //int transfer_type = pdlg->get_transfer_type();
        //if ( transfer_type == TransferThread::TRANSFER_GET )
        {
            // this->local_view->update_layout();
            // local 不需要吧,这个能自动更新的???
            this->update_layout();
        }
        //else if ( transfer_type == TransferThread::TRANSFER_PUT )
        {
            // remote_view->update_layout();
        }
        //else
        {
            // xxxxx: 没有预期到的错误
            //assert ( 1== 2 );
        }
    } else if (status == 52 // Transportor::ERROR_CANCEL
               ) {
        // user cancel, show nothing
    } else if (status != 0 // && status != 3
               ) {
        QString errmsg = QString(errorString + " Code: %1").arg(status);
        if (errmsg.length() < 50) errmsg = errmsg.leftJustified(50);
        QMessageBox::critical(this, QString(tr("Error: ")), errmsg);
    } else {
        Q_ASSERT(1 == 2);
    }
    // remote_view->slot_leave_remote_dir_retrive_loop();
}
コード例 #11
0
ファイル: httpwindow.cpp プロジェクト: coshaugh/CS441
void HttpWindow::startRequest(const QUrl &requestedUrl)
{
    url = requestedUrl;
    httpRequestAborted = false;

    reply = qnam.get(QNetworkRequest(url));
    connect(reply, &QNetworkReply::finished, this, &HttpWindow::httpFinished);
    connect(reply, &QIODevice::readyRead, this, &HttpWindow::httpReadyRead);

    ProgressDialog *progressDialog = new ProgressDialog(url, this);
    progressDialog->setAttribute(Qt::WA_DeleteOnClose);
    connect(progressDialog, &QProgressDialog::canceled, this, &HttpWindow::cancelDownload);
    connect(reply, &QNetworkReply::downloadProgress, progressDialog, &ProgressDialog::networkReplyProgress);
    connect(reply, &QNetworkReply::finished, progressDialog, &ProgressDialog::hide);
    progressDialog->show();

    statusLabel->setText(tr("Downloading %1...").arg(url.toString()));
    songBeingDownloaded = true;//
}
コード例 #12
0
ファイル: tags.cpp プロジェクト: JHooverman/phototonic
void ImageTags::applyUserAction(QList<QTreeWidgetItem *> tagsList)
{
	int processEventsCounter = 0;
	ProgressDialog *dialog = new ProgressDialog(this);
	dialog->show();

	QStringList currentSelectedImages = thumbView->getSelectedThumbsList();
	for (int i = 0; i < currentSelectedImages.size(); ++i) {

		QString imageName = currentSelectedImages[i]; 
		for (int i = tagsList.size() - 1; i > -1; --i) {
			Qt::CheckState tagState = tagsList.at(i)->checkState(0);
			setTagIcon(tagsList.at(i), (tagState == Qt::Checked? TagIconEnabled : TagIconDisabled));
			QString tagName = tagsList.at(i)->text(0);

			if (tagState == Qt::Checked) {
				dialog->opLabel->setText(tr("Tagging ") + imageName);
				mdCache->addTagToImage(imageName, tagName);
			} else {
				dialog->opLabel->setText(tr("Untagging ") + imageName);
				mdCache->removeTagFromImage(imageName, tagName);
			}
		}

		if (!writeTagsToImage(imageName, mdCache->getImageTags(imageName))) {
			mdCache->removeImage(imageName);
		}

		++processEventsCounter;
		if (processEventsCounter > 9) {
			processEventsCounter = 0;
			QApplication::processEvents();
		}

		if (dialog->abortOp) {
			break;
		}
	}

	dialog->close();
	delete(dialog);
}
コード例 #13
0
void ResourceManagementUtil::editResource( const std::string& path, const QIcon& resourceIcon )
{
   ProgressDialog progressDlg;
   progressDlg.initialize( "Loading a resource", 1 );

   ResourcesManager& resMgr = ResourcesManager::getInstance();
   Resource* resource = resMgr.create( path );

   progressDlg.advance();

   if ( resource )
   {
      TamyEditor& tamyEd = TamyEditor::getInstance();
      if ( tamyEd.activateResourceEditor( resource ) == false )
      {
         ResourceEditor* resourceEd = tamyEd.createResourceEditor( resource, resourceIcon );
         tamyEd.addResourceEditor( resourceEd );
      }
   }
}
コード例 #14
0
ファイル: ProgressDialog.cpp プロジェクト: ali-howie/zpack
bool ZpCallback(const zp::Char* path, zp::u32 fileSize, void* param)
{
	ProgressDialog* dlg = (ProgressDialog*)param;
	if (dlg == NULL)
	{
		return true;
	}
	if (dlg->m_running)
	{
		if (dlg->m_totalFileSize != 0)
		{
			dlg->setProgress(path, (float)dlg->m_doneFileSize / dlg->m_totalFileSize);
		}
		else
		{
			dlg->setProgress(path, 1.0f);
		}
	}
	dlg->m_doneFileSize += fileSize;
	return dlg->m_running;
}
コード例 #15
0
ファイル: FreqWindow.cpp プロジェクト: ruthmagnus/audacity
void FreqWindow::Recalc()
{
   wxLogMessage(wxT("Starting FreqWindow::Recalc()"));
   if (mProcessed)
      delete[] mProcessed;
   mProcessed = NULL;

   if (!mData) {
      mFreqPlot->Refresh(true);
      return;
   }

   int alg = mAlgChoice->GetSelection();
   int windowFunc = mFuncChoice->GetSelection();
   long windowSize = 0;
   (mSizeChoice->GetStringSelection()).ToLong(&windowSize);

   int f = NumWindowFuncs();

   if (!(windowSize >= 32 && windowSize <= 65536 &&
         alg >= 0 && alg <= 4 && windowFunc >= 0 && windowFunc < f)) {
      mFreqPlot->Refresh(true);
      return;
   }

   mWindowSize = windowSize;

   if (mDataLen < mWindowSize) {
      mFreqPlot->Refresh(true);
      return;
   }

   mProcessed = new float[mWindowSize];

   int i;
   for (i = 0; i < mWindowSize; i++)
      mProcessed[i] = float(0.0);
   int half = mWindowSize / 2;

   float *in = new float[mWindowSize];
   float *in2 = new float[mWindowSize];
   float *out = new float[mWindowSize];
   float *out2 = new float[mWindowSize];
   float *win = new float[mWindowSize];

   // initialize the window
   for(int i=0; i<mWindowSize; i++)
      win[i] = 1.0;
   WindowFunc(windowFunc, mWindowSize, win);
   // Scale window such that an amplitude of 1.0 in the time domain
   // shows an amplitude of 0dB in the frequency domain
   double wss = 0;
   for(int i=0; i<mWindowSize; i++)
      wss += win[i];
   if(wss > 0)
      wss = 4.0 / (wss*wss);
   else
      wss = 1.0;

   //Progress dialog over FFT operation
   wxLogMessage(wxT("Starting progress dialogue in FreqWindow::Recalc()"));
   ProgressDialog *mProgress = new ProgressDialog(_("FreqWindow"),_("Drawing Spectrum"));

   int start = 0;
   int windows = 0;
   while (start + mWindowSize <= mDataLen) {
      for (i = 0; i < mWindowSize; i++)
         in[i] = win[i] * mData[start + i];

      switch (alg) {
      case 0:                  // Spectrum
         PowerSpectrum(mWindowSize, in, out);

         for (i = 0; i < half; i++)
            mProcessed[i] += out[i];
         break;

      case 1:
      case 2:
      case 3:   // Autocorrelation, Cuberoot AC or Enhanced AC

         // Take FFT
#ifdef EXPERIMENTAL_USE_REALFFTF
         RealFFT(mWindowSize, in, out, out2);
#else
         FFT(mWindowSize, false, in, NULL, out, out2);
#endif
         // Compute power
         for (i = 0; i < mWindowSize; i++)
            in[i] = (out[i] * out[i]) + (out2[i] * out2[i]);

         if (alg == 1) {
            for (i = 0; i < mWindowSize; i++)
               in[i] = sqrt(in[i]);
         }
         if (alg == 2 || alg == 3) {
            // Tolonen and Karjalainen recommend taking the cube root
            // of the power, instead of the square root

            for (i = 0; i < mWindowSize; i++)
               in[i] = pow(in[i], 1.0f / 3.0f);
         }
         // Take FFT
#ifdef EXPERIMENTAL_USE_REALFFTF
         RealFFT(mWindowSize, in, out, out2);
#else
         FFT(mWindowSize, false, in, NULL, out, out2);
#endif

         // Take real part of result
         for (i = 0; i < half; i++)
            mProcessed[i] += out[i];
         break;

      case 4:                  // Cepstrum
#ifdef EXPERIMENTAL_USE_REALFFTF
         RealFFT(mWindowSize, in, out, out2);
#else
         FFT(mWindowSize, false, in, NULL, out, out2);
#endif

         // Compute log power
         // Set a sane lower limit assuming maximum time amplitude of 1.0
         float power;
         float minpower = 1e-20*mWindowSize*mWindowSize;
         for (i = 0; i < mWindowSize; i++)
         {
            power = (out[i] * out[i]) + (out2[i] * out2[i]);
            if(power < minpower)
               in[i] = log(minpower);
            else
               in[i] = log(power);
         }
         // Take IFFT
#ifdef EXPERIMENTAL_USE_REALFFTF
         InverseRealFFT(mWindowSize, in, NULL, out);
#else
         FFT(mWindowSize, true, in, NULL, out, out2);
#endif

         // Take real part of result
         for (i = 0; i < half; i++)
            mProcessed[i] += out[i];

         break;
      }                         //switch

      start += half;
      windows++;
      // only update the progress dialogue infrequently to reduce it's overhead
      // If we do it every time, it spends as much time updating X11 as doing
      // the calculations. 10 seems a reasonable compromise on Linux that
      // doesn't make it unresponsive, but avoids the slowdown.
      if ((windows % 10) == 0)
         mProgress->Update(1 - static_cast<float>(mDataLen - start) / mDataLen);
   }

   wxLogMessage(wxT("Finished updating progress dialogue in FreqWindow::Recalc()"));
   switch (alg) {
   double scale;
   case 0:                     // Spectrum
      // Convert to decibels
      mYMin = 1000000.;
      mYMax = -1000000.;
      scale = wss / (double)windows;
      for (i = 0; i < half; i++)
      {
         mProcessed[i] = 10 * log10(mProcessed[i] * scale);
         if(mProcessed[i] > mYMax)
            mYMax = mProcessed[i];
         else if(mProcessed[i] < mYMin)
            mYMin = mProcessed[i];
      }
      if(mYMin < -dBRange)
         mYMin = -dBRange;
      if(mYMax <= -dBRange)
         mYMax = -dBRange + 10.; // it's all out of range, but show a scale.
      else
         mYMax += .5;

      mProcessedSize = half;
      mYStep = 10;
      break;

   case 1:                     // Standard Autocorrelation
   case 2:                     // Cuberoot Autocorrelation
      for (i = 0; i < half; i++)
         mProcessed[i] = mProcessed[i] / windows;

      // Find min/max
      mYMin = mProcessed[0];
      mYMax = mProcessed[0];
      for (i = 1; i < half; i++)
         if (mProcessed[i] > mYMax)
            mYMax = mProcessed[i];
         else if (mProcessed[i] < mYMin)
            mYMin = mProcessed[i];

      mYStep = 1;

      mProcessedSize = half;
      break;

   case 3:                     // Enhanced Autocorrelation
      for (i = 0; i < half; i++)
         mProcessed[i] = mProcessed[i] / windows;

      // Peak Pruning as described by Tolonen and Karjalainen, 2000

      // Clip at zero, copy to temp array
      for (i = 0; i < half; i++) {
         if (mProcessed[i] < 0.0)
            mProcessed[i] = float(0.0);
         out[i] = mProcessed[i];
      }

      // Subtract a time-doubled signal (linearly interp.) from the original
      // (clipped) signal
      for (i = 0; i < half; i++)
         if ((i % 2) == 0)
            mProcessed[i] -= out[i / 2];
         else
            mProcessed[i] -= ((out[i / 2] + out[i / 2 + 1]) / 2);

      // Clip at zero again
      for (i = 0; i < half; i++)
         if (mProcessed[i] < 0.0)
            mProcessed[i] = float(0.0);

      // Find new min/max
      mYMin = mProcessed[0];
      mYMax = mProcessed[0];
      for (i = 1; i < half; i++)
         if (mProcessed[i] > mYMax)
            mYMax = mProcessed[i];
         else if (mProcessed[i] < mYMin)
            mYMin = mProcessed[i];

      mYStep = 1;

      mProcessedSize = half;
      break;

   case 4:                     // Cepstrum
      for (i = 0; i < half; i++)
         mProcessed[i] = mProcessed[i] / windows;

      // Find min/max, ignoring first and last few values
      int ignore = 4;
      mYMin = mProcessed[ignore];
      mYMax = mProcessed[ignore];
      for (i = ignore + 1; i < half - ignore; i++)
         if (mProcessed[i] > mYMax)
            mYMax = mProcessed[i];
         else if (mProcessed[i] < mYMin)
            mYMin = mProcessed[i];

      mYStep = 1;

      mProcessedSize = half;
      break;
   }

   delete[]in;
   delete[]in2;
   delete[]out;
   delete[]out2;
   delete[]win;

   wxLogMessage(wxT("About to draw plot in FreqWindow::Recalc()"));
   DrawPlot();
   mFreqPlot->Refresh(true);
   delete mProgress;
}
コード例 #16
0
ファイル: ExportOGG.cpp プロジェクト: ducknoir/audacity
int ExportOGG::Export(AudacityProject *project,
                       int numChannels,
                       wxString fName,
                       bool selectionOnly,
                       double t0,
                       double t1,
                       MixerSpec *mixerSpec,
                       Tags *metadata,
                       int WXUNUSED(subformat))
{
   double    rate    = project->GetRate();
   TrackList *tracks = project->GetTracks();
   double    quality = (gPrefs->Read(wxT("/FileFormats/OggExportQuality"), 50)/(float)100.0);

   wxLogNull logNo;            // temporarily disable wxWidgets error messages
   int updateResult = eProgressSuccess;
   int       eos = 0;

   FileIO outFile(fName, FileIO::Output);

   if (!outFile.IsOpened()) {
      wxMessageBox(_("Unable to open target file for writing"));
      return false;
   }

   // All the Ogg and Vorbis encoding data
   ogg_stream_state stream;
   ogg_page         page;
   ogg_packet       packet;

   vorbis_info      info;
   vorbis_comment   comment;
   vorbis_dsp_state dsp;
   vorbis_block     block;

   // Encoding setup
   vorbis_info_init(&info);
   vorbis_encode_init_vbr(&info, numChannels, int(rate + 0.5), quality);

   // Retrieve tags
   if (!FillComment(project, &comment, metadata)) {
      return false;
   }

   // Set up analysis state and auxiliary encoding storage
   vorbis_analysis_init(&dsp, &info);
   vorbis_block_init(&dsp, &block);

   // Set up packet->stream encoder.  According to encoder example,
   // a random serial number makes it more likely that you can make
   // chained streams with concatenation.
   srand(time(NULL));
   ogg_stream_init(&stream, rand());

   // First we need to write the required headers:
   //    1. The Ogg bitstream header, which contains codec setup params
   //    2. The Vorbis comment header
   //    3. The bitstream codebook.
   //
   // After we create those our responsibility is complete, libvorbis will
   // take care of any other ogg bistream constraints (again, according
   // to the example encoder source)
   ogg_packet bitstream_header;
   ogg_packet comment_header;
   ogg_packet codebook_header;

   vorbis_analysis_headerout(&dsp, &comment, &bitstream_header, &comment_header,
         &codebook_header);

   // Place these headers into the stream
   ogg_stream_packetin(&stream, &bitstream_header);
   ogg_stream_packetin(&stream, &comment_header);
   ogg_stream_packetin(&stream, &codebook_header);

   // Flushing these headers now guarentees that audio data will
   // start on a NEW page, which apparently makes streaming easier
   while (ogg_stream_flush(&stream, &page)) {
      outFile.Write(page.header, page.header_len);
      outFile.Write(page.body, page.body_len);
   }

   int numWaveTracks;
   WaveTrack **waveTracks;
   tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
   Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
                            tracks->GetTimeTrack(),
                            t0, t1,
                            numChannels, SAMPLES_PER_RUN, false,
                            rate, floatSample, true, mixerSpec);
   delete [] waveTracks;

   ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
      selectionOnly ?
      _("Exporting the selected audio as Ogg Vorbis") :
      _("Exporting the entire project as Ogg Vorbis"));

   while (updateResult == eProgressSuccess && !eos) {
      float **vorbis_buffer = vorbis_analysis_buffer(&dsp, SAMPLES_PER_RUN);
      sampleCount samplesThisRun = mixer->Process(SAMPLES_PER_RUN);

      if (samplesThisRun == 0) {
         // Tell the library that we wrote 0 bytes - signalling the end.
         vorbis_analysis_wrote(&dsp, 0);
      }
      else {

         for (int i = 0; i < numChannels; i++) {
            float *temp = (float *)mixer->GetBuffer(i);
            memcpy(vorbis_buffer[i], temp, sizeof(float)*SAMPLES_PER_RUN);
         }

         // tell the encoder how many samples we have
         vorbis_analysis_wrote(&dsp, samplesThisRun);
      }

      // I don't understand what this call does, so here is the comment
      // from the example, verbatim:
      //
      //    vorbis does some data preanalysis, then divvies up blocks
      //    for more involved (potentially parallel) processing. Get
      //    a single block for encoding now
      while (vorbis_analysis_blockout(&dsp, &block) == 1) {

         // analysis, assume we want to use bitrate management
         vorbis_analysis(&block, NULL);
         vorbis_bitrate_addblock(&block);

         while (vorbis_bitrate_flushpacket(&dsp, &packet)) {

            // add the packet to the bitstream
            ogg_stream_packetin(&stream, &packet);

            // From vorbis-tools-1.0/oggenc/encode.c:
            //   If we've gone over a page boundary, we can do actual output,
            //   so do so (for however many pages are available).

            while (!eos) {
               int result = ogg_stream_pageout(&stream, &page);
               if (!result) {
                  break;
               }

               outFile.Write(page.header, page.header_len);
               outFile.Write(page.body, page.body_len);

               if (ogg_page_eos(&page)) {
                  eos = 1;
               }
            }
         }
      }

      updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
   }

   delete progress;;

   delete mixer;

   ogg_stream_clear(&stream);

   vorbis_block_clear(&block);
   vorbis_dsp_clear(&dsp);
   vorbis_info_clear(&info);
   vorbis_comment_clear(&comment);

   outFile.Close();

   return updateResult;
}
コード例 #17
0
ファイル: ExportFFmpeg.cpp プロジェクト: MartynShaw/audacity
int ExportFFmpeg::Export(AudacityProject *project,
                       int channels, wxString fName,
                       bool selectionOnly, double t0, double t1, MixerSpec *mixerSpec, Tags *metadata, int subformat)
{
   if (!CheckFFmpegPresence())
      return false;
   mChannels = channels;
   // subformat index may not correspond directly to fmts[] index, convert it
   mSubFormat = AdjustFormatIndex(subformat);
   if (channels > ExportFFmpegOptions::fmts[mSubFormat].maxchannels)
   {
      wxMessageBox(
         wxString::Format(
               _("Attempted to export %d channels, but maximum number of channels for selected output format is %d"),
               channels,
               ExportFFmpegOptions::fmts[mSubFormat].maxchannels),
            _("Error"));
      return false;
   }
   mName = fName;
   TrackList *tracks = project->GetTracks();
   bool ret = true;

   if (mSubFormat >= FMT_LAST) return false;

   wxString shortname(ExportFFmpegOptions::fmts[mSubFormat].shortname);
   if (mSubFormat == FMT_OTHER)
      shortname = gPrefs->Read(wxT("/FileFormats/FFmpegFormat"),wxT("matroska"));
   ret = Init(shortname.mb_str(),project, metadata, subformat);

   if (!ret) return false;

   int pcmBufferSize = 1024;
   int numWaveTracks;
   WaveTrack **waveTracks;
   tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
   Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
      tracks->GetTimeTrack(),
      t0, t1,
      channels, pcmBufferSize, true,
      mSampleRate, int16Sample, true, mixerSpec);
   delete [] waveTracks;

   ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
      selectionOnly ?
      wxString::Format(_("Exporting selected audio as %s"), ExportFFmpegOptions::fmts[mSubFormat].description) :
   wxString::Format(_("Exporting entire file as %s"), ExportFFmpegOptions::fmts[mSubFormat].description));

   int updateResult = eProgressSuccess;

   while(updateResult == eProgressSuccess) {
      sampleCount pcmNumSamples = mixer->Process(pcmBufferSize);

      if (pcmNumSamples == 0)
         break;

      short *pcmBuffer = (short *)mixer->GetBuffer();

      EncodeAudioFrame(pcmBuffer,(pcmNumSamples)*sizeof(int16_t)*mChannels);

      updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
   }

   delete progress;

   delete mixer;

   Finalize();

   return updateResult;
}
コード例 #18
0
ファイル: VSTEffect.cpp プロジェクト: ruthmagnus/audacity
/* static */
void VSTEffect::Scan()
{
   wxArrayString audacityPathList = wxGetApp().audacityPathList;
   wxArrayString pathList;
   wxArrayString files;

   // Check for the VST_PATH environment variable
   wxString vstpath = wxGetenv(wxT("VST_PATH"));
   if (!vstpath.IsEmpty()) {
      wxGetApp().AddUniquePathToPathList(vstpath, pathList);
   }

   // Add Audacity specific paths
   for (size_t i = 0; i < audacityPathList.GetCount(); i++) {
      wxString prefix = audacityPathList[i] + wxFILE_SEP_PATH;
      wxGetApp().AddUniquePathToPathList(prefix + VSTPLUGINTYPE,
                                         pathList);
      wxGetApp().AddUniquePathToPathList(prefix + wxT("plugins"),
                                         pathList);
      wxGetApp().AddUniquePathToPathList(prefix + wxT("plug-ins"),
                                         pathList);
   }

#if defined(__WXMAC__)
#define VSTPATH wxT("/Library/Audio/Plug-Ins/VST")

   // Look in /Library/Audio/Plug-Ins/VST and $HOME/Library/Audio/Plug-Ins/VST
   wxGetApp().AddUniquePathToPathList(VSTPATH, pathList);
   wxGetApp().AddUniquePathToPathList(wxString(wxGetenv(wxT("HOME"))) + VSTPATH,
                                      pathList);

   // Recursively search all paths for Info.plist files.  This will identify all
   // bundles.
   wxGetApp().FindFilesInPathList(wxT("Info.plist"), pathList, files, wxDIR_DEFAULT);

   // Remove the 'Contents/Info.plist' portion of the names
   for (size_t i = 0, cnt = files.GetCount(); i < cnt; i++) {
      files[i] = wxPathOnly(wxPathOnly(files[i]));
   }
   
#elif defined(__WXMSW__)
   TCHAR dpath[MAX_PATH];
   TCHAR tpath[MAX_PATH];
   DWORD len = WXSIZEOF(tpath);

   // Setup the default VST path.
   dpath[0] = '\0';
   ExpandEnvironmentStrings(wxT("%ProgramFiles%\\Steinberg\\VSTPlugins"),
                            dpath,
                            WXSIZEOF(dpath));

   // Check registry for the real path
   if (SHRegGetUSValue(wxT("Software\\VST"),
                          wxT("VSTPluginsPath"),
                          NULL,
                          tpath,
                          &len,
                          FALSE,
                          dpath,
                          (DWORD) _tcslen(dpath)) == ERROR_SUCCESS) {
      tpath[len] = 0;
      ExpandEnvironmentStrings(tpath, dpath, WXSIZEOF(dpath));
      wxGetApp().AddUniquePathToPathList(LAT1CTOWX(dpath), pathList);
   }

   // Recursively scan for all DLLs
   wxGetApp().FindFilesInPathList(wxT("*.dll"), pathList, files, wxDIR_DEFAULT);

#else

   // Recursively scan for all shared objects
   wxGetApp().FindFilesInPathList(wxT("*.so"), pathList, files);

#endif

   // This is a hack to allow for long paths in the progress dialog.  The
   // progress dialog should really truncate the message if it's too wide
   // for the dialog.
   size_t cnt = files.GetCount();
   wxString longest;

   // JKC: Let's not show the progress dialog if there are no 
   // files to test.
   if( cnt <= 0 )
      return;

   for (size_t i = 0; i < cnt; i++) {
      if (files[i].Length() > longest.Length()) {
         longest = files[i];
      }
   }

   ProgressDialog *progress = new ProgressDialog(_("Scanning VST Plugins"),
                                                 longest,
                                                 pdlgHideStopButton);
//   progress->SetSize(wxSize(500, -1));
   progress->CenterOnScreen();

   const wxChar * argv[4];
   argv[0] = PlatformCompatibility::GetExecutablePath().c_str();
   argv[1] = VSTCMDKEY;
   argv[2] = NULL;
   argv[3] = NULL;

   for (size_t i = 0; i < cnt; i++) {
      wxString file = files[i];
      int status = progress->Update(wxLongLong(i),
                                    wxLongLong(cnt),
                                    wxString::Format(_("Checking %s"), file.c_str()));
      if (status != eProgressSuccess) {
         break;
      }

      argv[2] = file.c_str();
      // ToDo: do we need a try--catch around this in case a bad plug-in 
      // fails? (JKC Nov09)
      wxExecute((wxChar **) argv, wxEXEC_SYNC | wxEXEC_NODISABLE, NULL);
   }

   delete progress;   
}
コード例 #19
0
ファイル: ExportFLAC.cpp プロジェクト: jozsefmezei/audacity
int ExportFLAC::Export(AudacityProject *project,
                        int numChannels,
                        wxString fName,
                        bool selectionOnly,
                        double t0,
                        double t1,
                        MixerSpec *mixerSpec,
                        Tags *metadata,
                        int WXUNUSED(subformat))
{
   double    rate    = project->GetRate();
   TrackList *tracks = project->GetTracks();

   wxLogNull logNo;            // temporarily disable wxWidgets error messages
   int updateResult = eProgressSuccess;

   int levelPref;
   gPrefs->Read(wxT("/FileFormats/FLACLevel"), &levelPref, 5);

   wxString bitDepthPref =
      gPrefs->Read(wxT("/FileFormats/FLACBitDepth"), wxT("16"));

   FLAC::Encoder::File encoder;

#ifdef LEGACY_FLAC
   encoder.set_filename(OSOUTPUT(fName));
#endif
   encoder.set_channels(numChannels);
   encoder.set_sample_rate(lrint(rate));

   // See note in GetMetadata() about a bug in libflac++ 1.1.2
   if (!GetMetadata(project, metadata)) {
      return false;
   }

   if (mMetadata) {
      encoder.set_metadata(&mMetadata, 1);
   }

   sampleFormat format;
   if (bitDepthPref == wxT("24")) {
      format = int24Sample;
      encoder.set_bits_per_sample(24);
   } else { //convert float to 16 bits
      format = int16Sample;
      encoder.set_bits_per_sample(16);
   }

   // Duplicate the flac command line compression levels
   if (levelPref < 0 || levelPref > 8) {
      levelPref = 5;
   }
   encoder.set_do_exhaustive_model_search(flacLevels[levelPref].do_exhaustive_model_search);
   encoder.set_do_escape_coding(flacLevels[levelPref].do_escape_coding);
   if (numChannels != 2) {
      encoder.set_do_mid_side_stereo(false);
      encoder.set_loose_mid_side_stereo(false);
   }
   else {
      encoder.set_do_mid_side_stereo(flacLevels[levelPref].do_mid_side_stereo);
      encoder.set_loose_mid_side_stereo(flacLevels[levelPref].loose_mid_side_stereo);
   }
   encoder.set_qlp_coeff_precision(flacLevels[levelPref].qlp_coeff_precision);
   encoder.set_min_residual_partition_order(flacLevels[levelPref].min_residual_partition_order);
   encoder.set_max_residual_partition_order(flacLevels[levelPref].max_residual_partition_order);
   encoder.set_rice_parameter_search_dist(flacLevels[levelPref].rice_parameter_search_dist);
   encoder.set_max_lpc_order(flacLevels[levelPref].max_lpc_order);

#ifdef LEGACY_FLAC
   encoder.init();
#else
   wxFFile f;     // will be closed when it goes out of scope
   if (!f.Open(fName, wxT("w+b"))) {
      wxMessageBox(wxString::Format(_("FLAC export couldn't open %s"), fName.c_str()));
      return false;
   }

   // Even though there is an init() method that takes a filename, use the one that
   // takes a file handle because wxWidgets can open a file with a Unicode name and
   // libflac can't (under Windows).
   int status = encoder.init(f.fp());
   if (status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
      wxMessageBox(wxString::Format(_("FLAC encoder failed to initialize\nStatus: %d"), status));
      return false;
   }
#endif

   if (mMetadata) {
      ::FLAC__metadata_object_delete(mMetadata);
   }

   int numWaveTracks;
   WaveTrack **waveTracks;
   tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
   Mixer *mixer = CreateMixer(numWaveTracks, waveTracks,
                            tracks->GetTimeTrack(),
                            t0, t1,
                            numChannels, SAMPLES_PER_RUN, false,
                            rate, format, true, mixerSpec);
   delete [] waveTracks;

   int i, j;
   FLAC__int32 **tmpsmplbuf = new FLAC__int32*[numChannels];
   for (i = 0; i < numChannels; i++) {
      tmpsmplbuf[i] = (FLAC__int32 *) calloc(SAMPLES_PER_RUN, sizeof(FLAC__int32));
   }

   ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
         selectionOnly ?
         _("Exporting the selected audio as FLAC") :
         _("Exporting the entire project as FLAC"));

   while (updateResult == eProgressSuccess) {
      sampleCount samplesThisRun = mixer->Process(SAMPLES_PER_RUN);
      if (samplesThisRun == 0) { //stop encoding
         break;
      }
      else {
         for (i = 0; i < numChannels; i++) {
            samplePtr mixed = mixer->GetBuffer(i);
            if (format == int24Sample) {
               for (j = 0; j < samplesThisRun; j++) {
                  tmpsmplbuf[i][j] = ((int *) mixed)[j];
               }
            }
            else {
               for (j = 0; j < samplesThisRun; j++) {
                  tmpsmplbuf[i][j] = ((short *) mixed)[j];
               }
            }
         }
         encoder.process(tmpsmplbuf, samplesThisRun);
      }
      updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
   }
   f.Detach(); // libflac closes the file
   encoder.finish();

   delete progress;

   for (i = 0; i < numChannels; i++) {
      free(tmpsmplbuf[i]);
   }
   delete mixer;

   delete[] tmpsmplbuf;

   return updateResult;
}
コード例 #20
0
bool ExportFFmpeg::EncodeAudioFrame(int16_t *pFrame, int frameSize)
{
   AVPacket	pkt;
   int		nBytesToWrite = 0;
   uint8_t *	pRawSamples = NULL;
   int		nAudioFrameSizeOut = mEncAudioCodecCtx->frame_size * mEncAudioCodecCtx->channels * sizeof(int16_t);
   if (mEncAudioCodecCtx->frame_size == 1) nAudioFrameSizeOut = mEncAudioEncodedBufSiz;
   int      ret;

   nBytesToWrite = frameSize;
   pRawSamples  = (uint8_t*)pFrame;
#if FFMPEG_STABLE
   FFmpegLibsInst->av_fifo_realloc(&mEncAudioFifo, FFmpegLibsInst->av_fifo_size(&mEncAudioFifo) + frameSize);
#else
   FFmpegLibsInst->av_fifo_realloc2(mEncAudioFifo, FFmpegLibsInst->av_fifo_size(mEncAudioFifo) + frameSize);
#endif
   // Put the raw audio samples into the FIFO.
#if FFMPEG_STABLE
   ret = FFmpegLibsInst->av_fifo_generic_write(&mEncAudioFifo, pRawSamples, nBytesToWrite,NULL);
#else
   ret = FFmpegLibsInst->av_fifo_generic_write(mEncAudioFifo, pRawSamples, nBytesToWrite,NULL);
#endif
   wxASSERT(ret == nBytesToWrite);

   // Read raw audio samples out of the FIFO in nAudioFrameSizeOut byte-sized groups to encode.
#if FFMPEG_STABLE
   while ((ret = FFmpegLibsInst->av_fifo_size(&mEncAudioFifo)) >= nAudioFrameSizeOut)
   {
      ret = FFmpegLibsInst->av_fifo_read(&mEncAudioFifo, mEncAudioFifoOutBuf, nAudioFrameSizeOut);
#else
   while ((ret = FFmpegLibsInst->av_fifo_size(mEncAudioFifo)) >= nAudioFrameSizeOut)
   {
      ret = FFmpegLibsInst->av_fifo_generic_read(mEncAudioFifo, mEncAudioFifoOutBuf, nAudioFrameSizeOut, NULL);
#endif
      FFmpegLibsInst->av_init_packet(&pkt);

      pkt.size = FFmpegLibsInst->avcodec_encode_audio(mEncAudioCodecCtx, 
         mEncAudioEncodedBuf, mEncAudioEncodedBufSiz,		// out
         (int16_t*)mEncAudioFifoOutBuf);				// in
      if (mEncAudioCodecCtx->frame_size == 1) { wxASSERT(pkt.size == mEncAudioEncodedBufSiz); }
      if (pkt.size < 0)
      {
         wxLogMessage(wxT("FFmpeg : ERROR - Can't encode audio frame."));
         return false;
      }

      // Rescale from the codec time_base to the AVStream time_base.
      if (mEncAudioCodecCtx->coded_frame && mEncAudioCodecCtx->coded_frame->pts != int64_t(AV_NOPTS_VALUE))
         pkt.pts = FFmpegLibsInst->av_rescale_q(mEncAudioCodecCtx->coded_frame->pts, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
      //wxLogMessage(wxT("FFmpeg : (%d) Writing audio frame with PTS: %lld."), mEncAudioCodecCtx->frame_number, pkt.pts);

      pkt.stream_index = mEncAudioStream->index;
      pkt.data = mEncAudioEncodedBuf;
      pkt.flags |= PKT_FLAG_KEY;

      // Write the encoded audio frame to the output file.
      if ((ret = FFmpegLibsInst->av_interleaved_write_frame(mEncFormatCtx, &pkt)) != 0)
      {
         wxLogMessage(wxT("FFmpeg : ERROR - Failed to write audio frame to file."));
         return false;
      }
   }
   return true;
}


int ExportFFmpeg::Export(AudacityProject *project,
                       int channels, wxString fName,
                       bool selectionOnly, double t0, double t1, MixerSpec *mixerSpec, Tags *metadata, int subformat)
{
   if (!CheckFFmpegPresence())
      return false;
   mChannels = channels;
   // subformat index may not correspond directly to fmts[] index, convert it
   mSubFormat = AdjustFormatIndex(subformat);
   if (channels > ExportFFmpegOptions::fmts[mSubFormat].maxchannels)
   {
      wxLogMessage(wxT("Attempted to export %d channels, but max. channels = %d"),channels,ExportFFmpegOptions::fmts[mSubFormat].maxchannels);
      wxMessageBox(wxString::Format(_("Attempted to export %d channels, but max. channels for selected output format is %d"),channels,ExportFFmpegOptions::fmts[mSubFormat].maxchannels),_("Error"));
      return false;
   }
   mName = fName;
   TrackList *tracks = project->GetTracks();
   bool ret = true;

   if (mSubFormat >= FMT_LAST) return false;
   
   wxString shortname(ExportFFmpegOptions::fmts[mSubFormat].shortname);
   if (mSubFormat == FMT_OTHER)
      shortname = gPrefs->Read(wxT("/FileFormats/FFmpegFormat"),wxT("matroska"));
   ret = Init(shortname.mb_str(),project, metadata);

   if (!ret) return false;

   int pcmBufferSize = 1024;
   int numWaveTracks;
   WaveTrack **waveTracks;
   tracks->GetWaveTracks(selectionOnly, &numWaveTracks, &waveTracks);
   Mixer *mixer = new Mixer(numWaveTracks, waveTracks,
      tracks->GetTimeTrack(),
      t0, t1,
      channels, pcmBufferSize, true,
      mSampleRate, int16Sample, true, mixerSpec);
   delete [] waveTracks;

   ProgressDialog *progress = new ProgressDialog(wxFileName(fName).GetName(),
      selectionOnly ?
      wxString::Format(_("Exporting selected audio as %s"), ExportFFmpegOptions::fmts[mSubFormat].description) :
   wxString::Format(_("Exporting entire file as %s"), ExportFFmpegOptions::fmts[mSubFormat].description));

   int updateResult = eProgressSuccess;

   while(updateResult == eProgressSuccess) {
      sampleCount pcmNumSamples = mixer->Process(pcmBufferSize);

      if (pcmNumSamples == 0)
         break;

      short *pcmBuffer = (short *)mixer->GetBuffer();

      EncodeAudioFrame(pcmBuffer,(pcmNumSamples)*sizeof(int16_t)*mChannels);

      updateResult = progress->Update(mixer->MixGetCurrentTime()-t0, t1-t0);
   }

   delete progress;

   delete mixer;

   Finalize();

   return updateResult;
}

void AddStringTagUTF8(char field[], int size, wxString value)
{
      memset(field,0,size);
      memcpy(field,value.ToUTF8(),(int)strlen(value.ToUTF8()) > size -1 ? size -1 : strlen(value.ToUTF8()));
}
コード例 #21
0
// Given a project and a list of aliased files that should no
// longer be external dependencies (selected by the user), replace
// all of those alias block files with disk block files.
void RemoveDependencies(AudacityProject *project,
			               AliasedFileArray *aliasedFiles)
{
   DirManager *dirManager = project->GetDirManager();

   ProgressDialog *progress = 
      new ProgressDialog(_("Removing Dependencies"),
                         _("Copying audio data into project..."));
   int updateResult = eProgressSuccess;

   // Hash aliasedFiles based on their full paths and 
   // count total number of bytes to process.
   AliasedFileHash aliasedFileHash;
   wxLongLong totalBytesToProcess = 0;
   unsigned int i;
   for (i = 0; i < aliasedFiles->GetCount(); i++) {
      totalBytesToProcess += aliasedFiles->Item(i).mByteCount;
      wxString fileNameStr = aliasedFiles->Item(i).mFileName.GetFullPath();
      aliasedFileHash[fileNameStr] = &aliasedFiles->Item(i);
   }
   
   BlockArray blocks;
   GetAllSeqBlocks(project, &blocks);

   const sampleFormat format = project->GetDefaultFormat();
   ReplacedBlockFileHash blockFileHash;   
   wxLongLong completedBytes = 0;
   for (i = 0; i < blocks.GetCount(); i++) {
      BlockFile *f = blocks[i]->f;
      if (f->IsAlias() && (blockFileHash.count(f) == 0)) 
      {
         // f is an alias block we have not yet processed.
         AliasBlockFile *aliasBlockFile = (AliasBlockFile *)f;
         wxFileName fileName = aliasBlockFile->GetAliasedFileName();
         wxString fileNameStr = fileName.GetFullPath();

         if (aliasedFileHash.count(fileNameStr) == 0)
            // This aliased file was not selected to be replaced. Skip it.
            continue;

         // Convert it from an aliased file to an actual file in the project.
         unsigned int len = aliasBlockFile->GetLength();
         samplePtr buffer = NewSamples(len, format);
         f->ReadData(buffer, format, 0, len);
         BlockFile *newBlockFile =
            dirManager->NewSimpleBlockFile(buffer, len, format);
         DeleteSamples(buffer);

         // Update our hash so we know what block files we've done
         blockFileHash[f] = newBlockFile;

         // Update the progress bar
         completedBytes += SAMPLE_SIZE(format) * len;
         updateResult = progress->Update(completedBytes, totalBytesToProcess);
         if (updateResult != eProgressSuccess)
           break;
      }
   }

   // Above, we created a SimpleBlockFile contained in our project
   // to go with each AliasBlockFile that we wanted to migrate.
   // However, that didn't actually change any references to these
   // blockfiles in the Sequences, so we do that next...
   ReplaceBlockFiles(project, blockFileHash);

   // Subtract one from reference count of new block files; they're
   // now all referenced the proper number of times by the Sequences
   ReplacedBlockFileHash::iterator it;
   for( it = blockFileHash.begin(); it != blockFileHash.end(); ++it )
   {
      BlockFile *f = it->second;
      dirManager->Deref(f);
   }

   delete progress;
}
コード例 #22
0
ファイル: Mix.cpp プロジェクト: onuryuruten/audacity
//TODO-MB: wouldn't it make more sense to delete the time track after 'mix and render'?
bool MixAndRender(TrackList *tracks, TrackFactory *trackFactory,
                  double rate, sampleFormat format,
                  double startTime, double endTime,
                  WaveTrack **newLeft, WaveTrack **newRight)
{
   // This function was formerly known as "Quick Mix".
   WaveTrack **waveArray;
   Track *t;
   int numWaves = 0; /* number of wave tracks in the selection */
   int numMono = 0;  /* number of mono, centre-panned wave tracks in selection*/
   bool mono = false;   /* flag if output can be mono without loosing anything*/
   bool oneinput = false;  /* flag set to true if there is only one input track
                              (mono or stereo) */
   int w;

   TrackListIterator iter(tracks);
   SelectedTrackListOfKindIterator usefulIter(Track::Wave, tracks);
   // this only iterates tracks which are relevant to this function, i.e.
   // selected WaveTracks. The tracklist is (confusingly) the list of all
   // tracks in the project

   t = iter.First();
   while (t) {
      if (t->GetSelected() && t->GetKind() == Track::Wave) {
         numWaves++;
         float pan = ((WaveTrack*)t)->GetPan();
         if (t->GetChannel() == Track::MonoChannel && pan == 0)
            numMono++;
      }
      t = iter.Next();
   }

   if (numMono == numWaves)
      mono = true;

   /* the next loop will do two things at once:
    * 1. build an array of all the wave tracks were are trying to process
    * 2. determine when the set of WaveTracks starts and ends, in case we
    *    need to work out for ourselves when to start and stop rendering.
    */

   double mixStartTime = 0.0;    /* start time of first track to start */
   bool gotstart = false;  // flag indicates we have found a start time
   double mixEndTime = 0.0;   /* end time of last track to end */
   double tstart, tend;    // start and end times for one track.

   waveArray = new WaveTrack *[numWaves];
   w = 0;
   t = iter.First();

   while (t) {
      if (t->GetSelected() && t->GetKind() == Track::Wave) {
         waveArray[w++] = (WaveTrack *) t;
         tstart = t->GetStartTime();
         tend = t->GetEndTime();
         if (tend > mixEndTime)
            mixEndTime = tend;
         // try and get the start time. If the track is empty we will get 0,
         // which is ambiguous because it could just mean the track starts at
         // the beginning of the project, as well as empty track. The give-away
         // is that an empty track also ends at zero.

         if (tstart != tend) {
            // we don't get empty tracks here
            if (!gotstart) {
               // no previous start, use this one unconditionally
               mixStartTime = tstart;
               gotstart = true;
            } else if (tstart < mixStartTime)
               mixStartTime = tstart;  // have a start, only make it smaller
         }  // end if start and end are different
      }  // end if track is a selected WaveTrack.
      /** @TODO: could we not use a SelectedTrackListOfKindIterator here? */
      t = iter.Next();
   }

   /* create the destination track (new track) */
   if ((numWaves == 1) || ((numWaves == 2) && (usefulIter.First()->GetLink() != NULL)))
      oneinput = true;
   // only one input track (either 1 mono or one linked stereo pair)

   WaveTrack *mixLeft = trackFactory->NewWaveTrack(format, rate);
   if (oneinput)
      mixLeft->SetName(usefulIter.First()->GetName()); /* set name of output track to be the same as the sole input track */
   else
      mixLeft->SetName(_("Mix"));
   mixLeft->SetOffset(mixStartTime);
   WaveTrack *mixRight = 0;
   if (mono) {
      mixLeft->SetChannel(Track::MonoChannel);
   }
   else {
      mixRight = trackFactory->NewWaveTrack(format, rate);
      if (oneinput) {
         if (usefulIter.First()->GetLink() != NULL)   // we have linked track
            mixLeft->SetName(usefulIter.First()->GetLink()->GetName()); /* set name to match input track's right channel!*/
         else
            mixLeft->SetName(usefulIter.First()->GetName());   /* set name to that of sole input channel */
      }
      else
         mixRight->SetName(_("Mix"));
      mixLeft->SetChannel(Track::LeftChannel);
      mixRight->SetChannel(Track::RightChannel);
      mixRight->SetOffset(mixStartTime);
      mixLeft->SetLinked(true);
   }



   int maxBlockLen = mixLeft->GetIdealBlockSize();

   // If the caller didn't specify a time range, use the whole range in which
   // any input track had clips in it.
   if (startTime == endTime) {
      startTime = mixStartTime;
      endTime = mixEndTime;
   }

   Mixer *mixer = new Mixer(numWaves, waveArray,
                            Mixer::WarpOptions(tracks->GetTimeTrack()),
                            startTime, endTime, mono ? 1 : 2, maxBlockLen, false,
                            rate, format);

   ::wxSafeYield();
   ProgressDialog *progress = new ProgressDialog(_("Mix and Render"),
                                                 _("Mixing and rendering tracks"));

   int updateResult = eProgressSuccess;
   while(updateResult == eProgressSuccess) {
      sampleCount blockLen = mixer->Process(maxBlockLen);

      if (blockLen == 0)
         break;

      if (mono) {
         samplePtr buffer = mixer->GetBuffer();
         mixLeft->Append(buffer, format, blockLen);
      }
      else {
         samplePtr buffer;
         buffer = mixer->GetBuffer(0);
         mixLeft->Append(buffer, format, blockLen);
         buffer = mixer->GetBuffer(1);
         mixRight->Append(buffer, format, blockLen);
      }

      updateResult = progress->Update(mixer->MixGetCurrentTime() - startTime, endTime - startTime);
   }

   delete progress;

   mixLeft->Flush();
   if (!mono)
      mixRight->Flush();
   if (updateResult == eProgressCancelled || updateResult == eProgressFailed)
   {
      delete mixLeft;
      if (!mono)
         delete mixRight;
   } else {
      *newLeft = mixLeft;
      if (!mono)
         *newRight = mixRight;

#if 0
   int elapsedMS = wxGetElapsedTime();
   double elapsedTime = elapsedMS * 0.001;
   double maxTracks = totalTime / (elapsedTime / numWaves);

   // Note: these shouldn't be translated - they're for debugging
   // and profiling only.
   printf("      Tracks: %d\n", numWaves);
   printf("  Mix length: %f sec\n", totalTime);
   printf("Elapsed time: %f sec\n", elapsedTime);
   printf("Max number of tracks to mix in real time: %f\n", maxTracks);
#endif
   }

   delete[] waveArray;
   delete mixer;

   return (updateResult == eProgressSuccess || updateResult == eProgressStopped);
}