speed optimziation: use mono resampler for audio output
This commit is contained in:
parent
1c77e542aa
commit
6299f7ce92
5 changed files with 99 additions and 167 deletions
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@ -176,10 +176,11 @@ void CClient::Init()
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iMonoBlockSizeSam = MIN_BLOCK_SIZE_SAMPLES;
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iStereoBlockSizeSam = 2 * MIN_BLOCK_SIZE_SAMPLES;
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vecsAudioSndCrd.Init ( iSndCrdStereoBlockSizeSam );
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vecdAudioSndCrd.Init ( iSndCrdStereoBlockSizeSam );
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vecsAudioSndCrdStereo.Init ( iSndCrdStereoBlockSizeSam );
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vecdAudioSndCrdMono.Init ( iSndCrdMonoBlockSizeSam );
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vecdAudioSndCrdStereo.Init ( iSndCrdStereoBlockSizeSam );
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vecdAudio.Init ( iStereoBlockSizeSam );
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vecdAudioStereo.Init ( iStereoBlockSizeSam );
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Sound.InitRecording();
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Sound.InitPlayback();
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@ -250,7 +251,7 @@ void CClient::run()
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while ( bRun )
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{
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// get audio from sound card (blocking function)
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if ( Sound.Read ( vecsAudioSndCrd ) )
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if ( Sound.Read ( vecsAudioSndCrdStereo ) )
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{
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PostWinMessage ( MS_SOUND_IN, MUL_COL_LED_RED );
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}
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@ -262,14 +263,14 @@ void CClient::run()
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// convert data from short to double
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for ( i = 0; i < iSndCrdStereoBlockSizeSam; i++ )
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{
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vecdAudioSndCrd[i] = (double) vecsAudioSndCrd[i];
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vecdAudioSndCrdStereo[i] = (double) vecsAudioSndCrdStereo[i];
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}
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// resample data for each channel seaparately
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ResampleObjDown.Resample ( vecdAudioSndCrd, vecdAudio );
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ResampleObjDown.ResampleStereo ( vecdAudioSndCrdStereo, vecdAudioStereo );
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// update stereo signal level meter
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SignalLevelMeter.Update ( vecdAudio );
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SignalLevelMeter.Update ( vecdAudioStereo );
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// add reverberation effect if activated
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if ( iReverbLevel != 0 )
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@ -282,8 +283,8 @@ void CClient::run()
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for ( i = 0; i < iStereoBlockSizeSam; i += 2 )
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{
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// left channel
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vecdAudio[i] +=
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dRevLev * AudioReverb.ProcessSample ( vecdAudio[i] );
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vecdAudioStereo[i] +=
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dRevLev * AudioReverb.ProcessSample ( vecdAudioStereo[i] );
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}
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}
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else
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@ -291,25 +292,24 @@ void CClient::run()
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for ( i = 1; i < iStereoBlockSizeSam; i += 2 )
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{
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// right channel
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vecdAudio[i] +=
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dRevLev * AudioReverb.ProcessSample ( vecdAudio[i] );
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vecdAudioStereo[i] +=
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dRevLev * AudioReverb.ProcessSample ( vecdAudioStereo[i] );
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}
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}
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}
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// mix both signals depending on the fading setting
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const int iMiddleOfFader = AUD_FADER_IN_MAX / 2;
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const double dAttFact =
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(double) ( iMiddleOfFader - abs ( iMiddleOfFader - iAudioInFader ) ) /
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iMiddleOfFader;
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(double) ( AUD_FADER_IN_MIDDLE - abs ( AUD_FADER_IN_MIDDLE - iAudioInFader ) ) /
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AUD_FADER_IN_MIDDLE;
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if ( iAudioInFader > iMiddleOfFader )
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if ( iAudioInFader > AUD_FADER_IN_MIDDLE )
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{
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for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 )
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{
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// attenuation on right channel
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vecsNetwork[i] =
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Double2Short ( vecdAudio[j] + dAttFact * vecdAudio[j + 1] );
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Double2Short ( vecdAudioStereo[j] + dAttFact * vecdAudioStereo[j + 1] );
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}
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}
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else
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@ -318,7 +318,7 @@ void CClient::run()
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{
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// attenuation on left channel
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vecsNetwork[i] =
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Double2Short ( vecdAudio[j + 1] + dAttFact * vecdAudio[j] );
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Double2Short ( vecdAudioStereo[j + 1] + dAttFact * vecdAudioStereo[j] );
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}
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}
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@ -352,29 +352,25 @@ fflush(pFileDelay);
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// check if channel is connected
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if ( Channel.IsConnected() )
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{
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// write mono input signal in both sound-card channels
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for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 )
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// resample data
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ResampleObjUp.ResampleMono ( vecdNetwData, vecdAudioSndCrdMono );
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// convert data from double to short type and copy mono
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// received data in both sound card channels
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for ( i = 0, j = 0; i < iSndCrdMonoBlockSizeSam; i++, j += 2 )
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{
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vecdAudio[j] = vecdAudio[j + 1] = vecdNetwData[i];
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vecsAudioSndCrdStereo[j] = vecsAudioSndCrdStereo[j + 1] =
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Double2Short ( vecdAudioSndCrdMono[i] );
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}
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}
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else
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{
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// if not connected, clear data
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vecdAudio.Reset ( 0.0 );
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}
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// resample data
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ResampleObjUp.Resample ( vecdAudio, vecdAudioSndCrd );
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// convert data from double to short type
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for ( i = 0; i < iSndCrdStereoBlockSizeSam; i++ )
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{
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vecsAudioSndCrd[i] = Double2Short ( vecdAudioSndCrd[i] );
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vecsAudioSndCrdStereo.Reset ( 0 );
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}
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// play the new block
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if ( Sound.Write ( vecsAudioSndCrd ) )
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if ( Sound.Write ( vecsAudioSndCrdStereo ) )
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{
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PostWinMessage ( MS_SOUND_OUT, MUL_COL_LED_RED );
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}
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@ -49,7 +49,9 @@
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/* Definitions ****************************************************************/
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// audio in fader range
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#define AUD_FADER_IN_MIN 0
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#define AUD_FADER_IN_MAX 100
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#define AUD_FADER_IN_MIDDLE ( AUD_FADER_IN_MAX / 2 )
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// audio reverberation range
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#define AUD_REVERB_MAX 100
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@ -176,9 +178,10 @@ protected:
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bool bOpenChatOnNewMessage;
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CVector<short> vecsAudioSndCrd;
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CVector<double> vecdAudioSndCrd;
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CVector<double> vecdAudio;
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CVector<short> vecsAudioSndCrdStereo;
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CVector<double> vecdAudioSndCrdMono;
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CVector<double> vecdAudioSndCrdStereo;
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CVector<double> vecdAudioStereo;
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CVector<short> vecsNetwork;
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// resample objects
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@ -115,7 +115,7 @@ CLlconClientDlg::CLlconClientDlg ( CClient* pNCliP, QWidget* parent,
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// init slider controls ---
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// audio in fader
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SliderAudInFader->setRange ( 0, AUD_FADER_IN_MAX );
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SliderAudInFader->setRange ( AUD_FADER_IN_MIN, AUD_FADER_IN_MAX );
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const int iCurAudInFader = pClient->GetAudioInFader();
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SliderAudInFader->setValue ( iCurAudInFader );
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SliderAudInFader->setTickInterval ( AUD_FADER_IN_MAX / 9 );
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165
src/resample.cpp
165
src/resample.cpp
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@ -35,108 +35,10 @@
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#include "resample.h"
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/******************************************************************************\
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* General Resampler *
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\******************************************************************************/
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int CResample::Resample ( CVector<double>& vecdInput,
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CVector<double>& vecdOutput,
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const double dRation )
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{
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int i;
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/* move old data from the end to the history part of the buffer and
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add new data (shift register) */
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// shift old values
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int iMovLen = iInputBlockSize;
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for ( i = 0; i < iHistorySize; i++ )
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{
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vecdIntBuff[i] = vecdIntBuff[iMovLen++];
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}
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// add new block of data
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int iBlockEnd = iHistorySize;
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for ( i = 0; i < iInputBlockSize; i++ )
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{
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vecdIntBuff[iBlockEnd++] = vecdInput[i];
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}
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/* sample-interval of new sample frequency in relation to interpolated
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sample-interval */
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dTStep = (double) INTERP_DECIM_I_D1 / dRation;
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// init output counter
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int im = 0;
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// main loop
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do
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{
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// quantize output-time to interpolated time-index
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const int ik = (int) dtOut;
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/* Calculate convolutions for the two interpolation-taps ------------ */
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// phase for the linear interpolation-taps
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const int ip1 = ik % INTERP_DECIM_I_D1;
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const int ip2 = ( ik + 1 ) % INTERP_DECIM_I_D1;
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// sample positions in input vector
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const int in1 = (int) ( ik / INTERP_DECIM_I_D1 );
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const int in2 = (int) ( ( ik + 1 ) / INTERP_DECIM_I_D1 );
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// convolution
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double dy1 = 0.0;
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double dy2 = 0.0;
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for ( int i = 0; i < NUM_TAPS_PER_PHASE1; i++ )
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{
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dy1 += fResTaps1[ip1 * INTERP_DECIM_I_D1 + i] * vecdIntBuff[in1 - i];
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dy2 += fResTaps1[ip2 * INTERP_DECIM_I_D1 + i] * vecdIntBuff[in2 - i];
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}
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/* Linear interpolation --------------------------------------------- */
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// get numbers after the comma
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const double dxInt = dtOut - (int) dtOut;
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vecdOutput[im] = ( dy2 - dy1 ) * dxInt + dy1;
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// increase output counter
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im++;
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// increase output-time and index one step
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dtOut = dtOut + dTStep;
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}
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while ( dtOut < dBlockDuration );
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// set rtOut back
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dtOut -= iInputBlockSize * INTERP_DECIM_I_D1;
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return im;
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}
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void CResample::Init ( const int iNewInputBlockSize )
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{
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iInputBlockSize = iNewInputBlockSize;
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/* history size must be one sample larger, because we use always TWO
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convolutions */
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iHistorySize = NUM_TAPS_PER_PHASE1 + 1;
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// calculate block duration
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dBlockDuration = ( iInputBlockSize + iHistorySize - 1 ) * INTERP_DECIM_I_D1;
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// allocate memory for internal buffer, clear sample history
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vecdIntBuff.Init ( iInputBlockSize + iHistorySize, 0.0 );
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// init absolute time for output stream (at the end of the history part
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dtOut = (double) ( iHistorySize - 1 ) * INTERP_DECIM_I_D1;
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}
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/******************************************************************************\
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* Stereo Audio Resampler *
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\******************************************************************************/
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void CStereoAudioResample::Resample ( CVector<double>& vecdInput,
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void CStereoAudioResample::ResampleStereo ( CVector<double>& vecdInput,
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CVector<double>& vecdOutput )
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{
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int j;
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int iMovLen = iStereoInputBlockSize;
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for ( j = 0; j < iTwoTimesNumTaps; j++ )
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{
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vecdIntBuff[j] = vecdIntBuff[iMovLen++];
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vecdIntBuffStereo[j] = vecdIntBuffStereo[iMovLen++];
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}
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// add new block of data
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int iBlockEnd = iTwoTimesNumTaps;
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for ( j = 0; j < iStereoInputBlockSize; j++ )
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{
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vecdIntBuff[iBlockEnd++] = vecdInput[j];
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vecdIntBuffStereo[iBlockEnd++] = vecdInput[j];
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}
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// main loop
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@ -183,8 +85,8 @@ void CStereoAudioResample::Resample ( CVector<double>& vecdInput,
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const double dCurFiltTap = pFiltTaps[ip + i * iI];
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const int iCurSamplePos = in - 2 * i;
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dyL += dCurFiltTap * vecdIntBuff[iCurSamplePos];
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dyR += dCurFiltTap * vecdIntBuff[iCurSamplePos + 1];
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dyL += dCurFiltTap * vecdIntBuffStereo[iCurSamplePos];
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dyR += dCurFiltTap * vecdIntBuffStereo[iCurSamplePos + 1];
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}
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vecdOutput[2 * j] = dyL;
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@ -193,11 +95,61 @@ void CStereoAudioResample::Resample ( CVector<double>& vecdInput,
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}
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}
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void CStereoAudioResample::ResampleMono ( CVector<double>& vecdInput,
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CVector<double>& vecdOutput )
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{
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int j;
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if ( dRation == 1.0 )
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{
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// if ratio is 1, no resampling is needed, just copy vector
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vecdOutput = vecdInput;
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}
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else
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{
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/* move old data from the end to the history part of the buffer and
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add new data (shift register) */
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// shift old values
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int iMovLen = iMonoInputBlockSize;
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for ( j = 0; j < iNumTaps; j++ )
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{
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vecdIntBuffMono[j] = vecdIntBuffMono[iMovLen++];
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}
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// add new block of data
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int iBlockEnd = iNumTaps;
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for ( j = 0; j < iMonoInputBlockSize; j++ )
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{
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vecdIntBuffMono[iBlockEnd++] = vecdInput[j];
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}
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// main loop
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for ( j = 0; j < iMonoOutputBlockSize; j++ )
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{
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// calculate filter phase
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const int ip = (int) ( j * iI / dRation ) % iI;
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// sample position in input vector
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const int in = (int) ( j / dRation ) + iNumTaps - 1;
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// convolution
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double dy = 0.0;
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for ( int i = 0; i < iNumTaps; i++ )
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{
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dy += pFiltTaps[ip + i * iI] * vecdIntBuffMono[in - i];
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}
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vecdOutput[j] = dy;
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}
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}
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}
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void CStereoAudioResample::Init ( const int iNewMonoInputBlockSize,
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const int iFrom,
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const int iTo )
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{
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dRation = ( (double) iTo ) / iFrom;
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iMonoInputBlockSize = iNewMonoInputBlockSize;
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iStereoInputBlockSize = 2 * iNewMonoInputBlockSize;
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iMonoOutputBlockSize = (int) ( iNewMonoInputBlockSize * dRation );
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}
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}
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// allocate memory for internal buffer, clear sample history (we have
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// to consider stereo data here -> two times the number of taps of
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// allocate memory for internal buffer, clear sample history (for
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// the stereo case we have to consider that two times the number of taps of
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// additional memory is required)
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vecdIntBuff.Init ( iStereoInputBlockSize + 2 * iNumTaps, 0.0 );
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vecdIntBuffMono.Init ( iMonoInputBlockSize + iNumTaps, 0.0 );
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vecdIntBuffStereo.Init ( iStereoInputBlockSize + 2 * iNumTaps, 0.0 );
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}
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@ -31,27 +31,6 @@
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/* Classes ********************************************************************/
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class CResample
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{
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public:
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CResample() {}
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virtual ~CResample() {}
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void Init ( const int iNewInputBlockSize );
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int Resample ( CVector<double>& vecdInput, CVector<double>& vecdOutput,
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const double dRation );
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protected:
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double dTStep;
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double dtOut;
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double dBlockDuration;
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CVector<double> vecdIntBuff;
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int iHistorySize;
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int iInputBlockSize;
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};
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class CStereoAudioResample
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{
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public:
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virtual ~CStereoAudioResample() {}
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void Init ( const int iNewMonoInputBlockSize, const int iFrom, const int iTo );
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void Resample ( CVector<double>& vecdInput, CVector<double>& vecdOutput );
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void ResampleMono ( CVector<double>& vecdInput, CVector<double>& vecdOutput );
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void ResampleStereo ( CVector<double>& vecdInput, CVector<double>& vecdOutput );
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protected:
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double dRation;
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CVector<double> vecdIntBuff;
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int iHistorySize;
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CVector<double> vecdIntBuffMono;
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CVector<double> vecdIntBuffStereo;
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int iMonoInputBlockSize;
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int iStereoInputBlockSize;
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int iMonoOutputBlockSize;
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int iI;
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};
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#endif // !defined ( RESAMPLE_H__3B0FEUFE7876F_FE8FE_CA63_4344_1912__INCLUDED_ )
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