bool Sampler::__render_note_resample(
Sample *pSample,
Note *pNote,
SelectedLayerInfo *pSelectedLayerInfo,
InstrumentComponent *pCompo,
DrumkitComponent *pDrumCompo,
int nBufferSize,
int nInitialSilence,
float cost_L,
float cost_R,
float cost_track_L,
float cost_track_R,
float fLayerPitch,
Song* pSong
)
{
AudioOutput* pAudioOutput = Hydrogen::get_instance()->getAudioOutput();
int nNoteLength = -1;
if ( pNote->get_length() != -1 ) {
nNoteLength = ( int )( pNote->get_length() * pAudioOutput->m_transport.m_nTickSize );
}
float fNotePitch = pNote->get_total_pitch() + fLayerPitch;
float fStep = pow( 1.0594630943593, ( double )fNotePitch );
// _ERRORLOG( QString("pitch: %1, step: %2" ).arg(fNotePitch).arg( fStep) );
fStep *= ( float )pSample->get_sample_rate() / pAudioOutput->getSampleRate(); // Adjust for audio driver sample rate
// verifico il numero di frame disponibili ancora da eseguire
int nAvail_bytes = ( int )( ( float )( pSample->get_frames() - pSelectedLayerInfo->SamplePosition ) / fStep );
bool retValue = true; // the note is ended
if ( nAvail_bytes > nBufferSize - nInitialSilence ) { // il sample e' piu' grande del buffersize
// imposto il numero dei bytes disponibili uguale al buffersize
nAvail_bytes = nBufferSize - nInitialSilence;
retValue = false; // the note is not ended yet
}
// ADSR *pADSR = pNote->m_pADSR;
int nInitialBufferPos = nInitialSilence;
//float fInitialSamplePos = pNote->get_sample_position( pCompo->get_drumkit_componentID() );
double fSamplePos = pSelectedLayerInfo->SamplePosition;
int nTimes = nInitialBufferPos + nAvail_bytes;
float *pSample_data_L = pSample->get_data_l();
float *pSample_data_R = pSample->get_data_r();
float fInstrPeak_L = pNote->get_instrument()->get_peak_l(); // this value will be reset to 0 by the mixer..
float fInstrPeak_R = pNote->get_instrument()->get_peak_r(); // this value will be reset to 0 by the mixer..
float fADSRValue = 1.0;
float fVal_L;
float fVal_R;
int nSampleFrames = pSample->get_frames();
#ifdef H2CORE_HAVE_JACK
JackAudioDriver* pJackAudioDriver = 0;
float * pTrackOutL = 0;
float * pTrackOutR = 0;
if( pAudioOutput->has_track_outs()
&& (pJackAudioDriver = dynamic_cast<JackAudioDriver*>(pAudioOutput)) ) {
pTrackOutL = pJackAudioDriver->getTrackOut_L( pNote->get_instrument(), pCompo );
pTrackOutR = pJackAudioDriver->getTrackOut_R( pNote->get_instrument(), pCompo );
}
#endif
for ( int nBufferPos = nInitialBufferPos; nBufferPos < nTimes; ++nBufferPos ) {
if ( ( nNoteLength != -1 ) && ( nNoteLength <= pSelectedLayerInfo->SamplePosition ) ) {
if ( pNote->get_adsr()->release() == 0 ) {
retValue = 1; // the note is ended
}
}
int nSamplePos = ( int )fSamplePos;
double fDiff = fSamplePos - nSamplePos;
if ( ( nSamplePos + 1 ) >= nSampleFrames ) {
//we reach the last audioframe.
//set this last frame to zero do nothin wrong.
fVal_L = 0.0;
fVal_R = 0.0;
} else {
// some interpolation methods need 4 frames data.
float last_l;
float last_r;
if ( ( nSamplePos + 2 ) >= nSampleFrames ) {
last_l = 0.0;
last_r = 0.0;
} else {
last_l = pSample_data_L[nSamplePos + 2];
last_r = pSample_data_R[nSamplePos + 2];
}
switch( __interpolateMode ){
case LINEAR:
fVal_L = pSample_data_L[nSamplePos] * (1 - fDiff ) + pSample_data_L[nSamplePos + 1] * fDiff;
fVal_R = pSample_data_R[nSamplePos] * (1 - fDiff ) + pSample_data_R[nSamplePos + 1] * fDiff;
//fVal_L = linear_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
//fVal_R = linear_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case COSINE:
fVal_L = cosine_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
fVal_R = cosine_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case THIRD:
fVal_L = third_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = third_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case CUBIC:
fVal_L = cubic_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = cubic_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case HERMITE:
fVal_L = hermite_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = hermite_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
}
}
// ADSR envelope
fADSRValue = pNote->get_adsr()->get_value( fStep );
fVal_L = fVal_L * fADSRValue;
fVal_R = fVal_R * fADSRValue;
// Low pass resonant filter
if ( pNote->get_instrument()->is_filter_active() ) {
pNote->compute_lr_values( &fVal_L, &fVal_R );
}
#ifdef H2CORE_HAVE_JACK
if( pTrackOutL ) {
pTrackOutL[nBufferPos] += fVal_L * cost_track_L;
}
if( pTrackOutR ) {
pTrackOutR[nBufferPos] += fVal_R * cost_track_R;
}
#endif
fVal_L = fVal_L * cost_L;
fVal_R = fVal_R * cost_R;
// update instr peak
if ( fVal_L > fInstrPeak_L ) {
fInstrPeak_L = fVal_L;
}
if ( fVal_R > fInstrPeak_R ) {
fInstrPeak_R = fVal_R;
}
pDrumCompo->set_outs( nBufferPos, fVal_L, fVal_R );
// to main mix
__main_out_L[nBufferPos] += fVal_L;
__main_out_R[nBufferPos] += fVal_R;
fSamplePos += fStep;
}
pSelectedLayerInfo->SamplePosition += nAvail_bytes * fStep;
pNote->get_instrument()->set_peak_l( fInstrPeak_L );
pNote->get_instrument()->set_peak_r( fInstrPeak_R );
#ifdef H2CORE_HAVE_LADSPA
// LADSPA
// change the below return logic if you add code after that ifdef
if (pNote->get_instrument()->is_muted() || pSong->__is_muted) return retValue;
float masterVol = pSong->get_volume();
for ( unsigned nFX = 0; nFX < MAX_FX; ++nFX ) {
LadspaFX *pFX = Effects::get_instance()->getLadspaFX( nFX );
float fLevel = pNote->get_instrument()->get_fx_level( nFX );
if ( ( pFX ) && ( fLevel != 0.0 ) ) {
fLevel = fLevel * pFX->getVolume();
float *pBuf_L = pFX->m_pBuffer_L;
float *pBuf_R = pFX->m_pBuffer_R;
// float fFXCost_L = cost_L * fLevel;
// float fFXCost_R = cost_R * fLevel;
float fFXCost_L = fLevel * masterVol;
float fFXCost_R = fLevel * masterVol;
int nBufferPos = nInitialBufferPos;
float fSamplePos = pSelectedLayerInfo->SamplePosition;
for ( int i = 0; i < nAvail_bytes; ++i ) {
int nSamplePos = ( int )fSamplePos;
double fDiff = fSamplePos - nSamplePos;
if ( ( nSamplePos + 1 ) >= nSampleFrames ) {
//we reach the last audioframe.
//set this last frame to zero do nothin wrong.
fVal_L = 0.0;
fVal_R = 0.0;
} else {
// some interpolation methods need 4 frames data.
float last_l;
float last_r;
if ( ( nSamplePos + 2 ) >= nSampleFrames ) {
last_l = 0.0;
last_r = 0.0;
}else
{
last_l = pSample_data_L[nSamplePos + 2];
last_r = pSample_data_R[nSamplePos + 2];
}
switch( __interpolateMode ){
case LINEAR:
fVal_L = pSample_data_L[nSamplePos] * (1 - fDiff ) + pSample_data_L[nSamplePos + 1] * fDiff;
fVal_R = pSample_data_R[nSamplePos] * (1 - fDiff ) + pSample_data_R[nSamplePos + 1] * fDiff;
break;
case COSINE:
fVal_L = cosine_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
fVal_R = cosine_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case THIRD:
fVal_L = third_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = third_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case CUBIC:
fVal_L = cubic_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = cubic_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case HERMITE:
fVal_L = hermite_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = hermite_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
}
}
pBuf_L[ nBufferPos ] += fVal_L * fFXCost_L;
pBuf_R[ nBufferPos ] += fVal_R * fFXCost_R;
fSamplePos += fStep;
++nBufferPos;
}
}
}
#endif
return retValue;
}
int Sampler::__render_note_resample(
Sample *pSample,
Note *pNote,
int nBufferSize,
int nInitialSilence,
float cost_L,
float cost_R,
float cost_track_L,
float cost_track_R,
float fLayerPitch,
Song* pSong
)
{
AudioOutput* audio_output = Hydrogen::get_instance()->getAudioOutput();
int nNoteLength = -1;
if ( pNote->get_length() != -1 ) {
nNoteLength = ( int )( pNote->get_length() * audio_output->m_transport.m_nTickSize );
}
float fNotePitch = pNote->get_total_pitch() + fLayerPitch;
float fStep = pow( 1.0594630943593, ( double )fNotePitch );
// _ERRORLOG( QString("pitch: %1, step: %2" ).arg(fNotePitch).arg( fStep) );
fStep *= ( float )pSample->get_sample_rate() / audio_output->getSampleRate(); // Adjust for audio driver sample rate
// verifico il numero di frame disponibili ancora da eseguire
int nAvail_bytes = ( int )( ( float )( pSample->get_frames() - pNote->get_sample_position() ) / fStep );
int retValue = 1; // the note is ended
if ( nAvail_bytes > nBufferSize - nInitialSilence ) { // il sample e' piu' grande del buffersize
// imposto il numero dei bytes disponibili uguale al buffersize
nAvail_bytes = nBufferSize - nInitialSilence;
retValue = 0; // the note is not ended yet
}
// ADSR *pADSR = pNote->m_pADSR;
int nInitialBufferPos = nInitialSilence;
float fInitialSamplePos = pNote->get_sample_position();
double fSamplePos = pNote->get_sample_position();
int nTimes = nInitialBufferPos + nAvail_bytes;
int nInstrument = pSong->get_instrument_list()->index( pNote->get_instrument() );
float *pSample_data_L = pSample->get_data_l();
float *pSample_data_R = pSample->get_data_r();
float fInstrPeak_L = pNote->get_instrument()->get_peak_l(); // this value will be reset to 0 by the mixer..
float fInstrPeak_R = pNote->get_instrument()->get_peak_r(); // this value will be reset to 0 by the mixer..
float fADSRValue = 1.0;
float fVal_L;
float fVal_R;
int nSampleFrames = pSample->get_frames();
/*
* nInstrument could be -1 if the instrument is not found in the current drumset.
* This happens when someone is using the prelistening function of the soundlibrary.
*/
if( nInstrument < 0 ) {
nInstrument = 0;
}
#ifdef H2CORE_HAVE_JACK
JackOutput* jao = 0;
float *track_out_L = 0;
float *track_out_R = 0;
if( audio_output->has_track_outs()
&& (jao = dynamic_cast<JackOutput*>(audio_output)) ) {
track_out_L = jao->getTrackOut_L( nInstrument );
track_out_R = jao->getTrackOut_R( nInstrument );
}
#endif
for ( int nBufferPos = nInitialBufferPos; nBufferPos < nTimes; ++nBufferPos ) {
if ( ( nNoteLength != -1 ) && ( nNoteLength <= pNote->get_sample_position() ) ) {
if ( pNote->get_adsr()->release() == 0 ) {
retValue = 1; // the note is ended
}
}
int nSamplePos = ( int )fSamplePos;
double fDiff = fSamplePos - nSamplePos;
if ( ( nSamplePos + 1 ) >= nSampleFrames ) {
//we reach the last audioframe.
//set this last frame to zero do nothin wrong.
fVal_L = 0.0;
fVal_R = 0.0;
} else {
// some interpolation methods need 4 frames data.
float last_l;
float last_r;
if ( ( nSamplePos + 2 ) >= nSampleFrames ) {
last_l = 0.0;
last_r = 0.0;
} else {
last_l = pSample_data_L[nSamplePos + 2];
last_r = pSample_data_R[nSamplePos + 2];
}
switch( __interpolateMode ){
case LINEAR:
fVal_L = pSample_data_L[nSamplePos] * (1 - fDiff ) + pSample_data_L[nSamplePos + 1] * fDiff;
fVal_R = pSample_data_R[nSamplePos] * (1 - fDiff ) + pSample_data_R[nSamplePos + 1] * fDiff;
//fVal_L = linear_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
//fVal_R = linear_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case COSINE:
fVal_L = cosine_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
fVal_R = cosine_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case THIRD:
fVal_L = third_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = third_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case CUBIC:
fVal_L = cubic_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = cubic_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case HERMITE:
fVal_L = hermite_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = hermite_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
}
}
// ADSR envelope
fADSRValue = pNote->get_adsr()->get_value( fStep );
fVal_L = fVal_L * fADSRValue;
fVal_R = fVal_R * fADSRValue;
// Low pass resonant filter
if ( pNote->get_instrument()->is_filter_active() ) {
pNote->compute_lr_values( &fVal_L, &fVal_R );
}
#ifdef H2CORE_HAVE_JACK
if( track_out_L ) {
track_out_L[nBufferPos] += fVal_L * cost_track_L;
}
if( track_out_R ) {
track_out_R[nBufferPos] += fVal_R * cost_track_R;
}
#endif
fVal_L = fVal_L * cost_L;
fVal_R = fVal_R * cost_R;
// update instr peak
if ( fVal_L > fInstrPeak_L ) {
fInstrPeak_L = fVal_L;
}
if ( fVal_R > fInstrPeak_R ) {
fInstrPeak_R = fVal_R;
}
// to main mix
__main_out_L[nBufferPos] += fVal_L;
__main_out_R[nBufferPos] += fVal_R;
fSamplePos += fStep;
}
pNote->update_sample_position( nAvail_bytes * fStep );
pNote->get_instrument()->set_peak_l( fInstrPeak_L );
pNote->get_instrument()->set_peak_r( fInstrPeak_R );
#ifdef H2CORE_HAVE_LADSPA
// LADSPA
float masterVol = pSong->get_volume();
for ( unsigned nFX = 0; nFX < MAX_FX; ++nFX ) {
LadspaFX *pFX = Effects::get_instance()->getLadspaFX( nFX );
float fLevel = pNote->get_instrument()->get_fx_level( nFX );
if ( ( pFX ) && ( fLevel != 0.0 ) ) {
fLevel = fLevel * pFX->getVolume();
float *pBuf_L = pFX->m_pBuffer_L;
float *pBuf_R = pFX->m_pBuffer_R;
// float fFXCost_L = cost_L * fLevel;
// float fFXCost_R = cost_R * fLevel;
float fFXCost_L = fLevel * masterVol;
float fFXCost_R = fLevel * masterVol;
int nBufferPos = nInitialBufferPos;
float fSamplePos = fInitialSamplePos;
for ( int i = 0; i < nAvail_bytes; ++i ) {
int nSamplePos = ( int )fSamplePos;
double fDiff = fSamplePos - nSamplePos;
if ( ( nSamplePos + 1 ) >= nSampleFrames ) {
//we reach the last audioframe.
//set this last frame to zero do nothin wrong.
fVal_L = 0.0;
fVal_R = 0.0;
} else {
// some interpolation methods need 4 frames data.
float last_l;
float last_r;
if ( ( nSamplePos + 2 ) >= nSampleFrames ) {
last_l = 0.0;
last_r = 0.0;
}else
{
last_l = pSample_data_L[nSamplePos + 2];
last_r = pSample_data_R[nSamplePos + 2];
}
switch( __interpolateMode ){
case LINEAR:
fVal_L = pSample_data_L[nSamplePos] * (1 - fDiff ) + pSample_data_L[nSamplePos + 1] * fDiff;
fVal_R = pSample_data_R[nSamplePos] * (1 - fDiff ) + pSample_data_R[nSamplePos + 1] * fDiff;
//fVal_L = linear_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
//fVal_R = linear_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case COSINE:
fVal_L = cosine_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
fVal_R = cosine_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case THIRD:
fVal_L = third_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = third_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case CUBIC:
fVal_L = cubic_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = cubic_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case HERMITE:
fVal_L = hermite_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = hermite_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
}
// methode Interpolate produce an extra function call and eat much more time here.
// so i deside to code the switch direct in the resampler methode
//fVal_L = Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], pSample_data_L[nSamplePos + 2] ,fDiff);
//fVal_L = Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], pSample_data_R[nSamplePos + 2] ,fDiff);
}
pBuf_L[ nBufferPos ] += fVal_L * fFXCost_L;
pBuf_R[ nBufferPos ] += fVal_R * fFXCost_R;
fSamplePos += fStep;
++nBufferPos;
}
}
}
#endif
return retValue;
}
bool Sampler::processPlaybackTrack(int nBufferSize)
{
Hydrogen* pEngine = Hydrogen::get_instance();
AudioOutput* pAudioOutput = Hydrogen::get_instance()->getAudioOutput();
Song* pSong = pEngine->getSong();
if( !pSong->get_playback_track_enabled()
|| pEngine->getState() != STATE_PLAYING
|| pSong->get_mode() != Song::SONG_MODE)
{
return false;
}
InstrumentComponent *pCompo = __playback_instrument->get_components()->front();
Sample *pSample = pCompo->get_layer(0)->get_sample();
float fVal_L;
float fVal_R;
float *pSample_data_L = pSample->get_data_l();
float *pSample_data_R = pSample->get_data_r();
float fInstrPeak_L = __playback_instrument->get_peak_l(); // this value will be reset to 0 by the mixer..
float fInstrPeak_R = __playback_instrument->get_peak_r(); // this value will be reset to 0 by the mixer..
assert(pSample);
int nAvail_bytes = 0;
int nInitialBufferPos = 0;
if(pSample->get_sample_rate() == pAudioOutput->getSampleRate()){
//No resampling
__playBackSamplePosition = pAudioOutput->m_transport.m_nFrames;
nAvail_bytes = pSample->get_frames() - ( int )__playBackSamplePosition;
if ( nAvail_bytes > nBufferSize ) {
nAvail_bytes = nBufferSize;
}
int nInitialSamplePos = ( int ) __playBackSamplePosition;
int nSamplePos = nInitialSamplePos;
int nTimes = nInitialBufferPos + nAvail_bytes;
if(__playBackSamplePosition > pSample->get_frames()){
//playback track has ended..
return true;
}
for ( int nBufferPos = nInitialBufferPos; nBufferPos < nTimes; ++nBufferPos ) {
fVal_L = pSample_data_L[ nSamplePos ];
fVal_R = pSample_data_R[ nSamplePos ];
fVal_L = fVal_L * 1.0f * pSong->get_playback_track_volume(); //costr
fVal_R = fVal_R * 1.0f * pSong->get_playback_track_volume(); //cost l
//pDrumCompo->set_outs( nBufferPos, fVal_L, fVal_R );
// to main mix
if ( fVal_L > fInstrPeak_L ) {
fInstrPeak_L = fVal_L;
}
if ( fVal_R > fInstrPeak_R ) {
fInstrPeak_R = fVal_R;
}
__main_out_L[nBufferPos] += fVal_L;
__main_out_R[nBufferPos] += fVal_R;
++nSamplePos;
}
} else {
//Perform resampling
double fSamplePos = 0;
int nSampleFrames = pSample->get_frames();
float fStep = 1.0594630943593;
fStep *= ( float )pSample->get_sample_rate() / pAudioOutput->getSampleRate(); // Adjust for audio driver sample rate
if(pAudioOutput->m_transport.m_nFrames == 0){
fSamplePos = 0;
} else {
fSamplePos = ( (pAudioOutput->m_transport.m_nFrames/nBufferSize) * (nBufferSize * fStep));
}
nAvail_bytes = ( int )( ( float )( pSample->get_frames() - fSamplePos ) / fStep );
if ( nAvail_bytes > nBufferSize ) {
nAvail_bytes = nBufferSize;
}
int nTimes = nInitialBufferPos + nAvail_bytes;
for ( int nBufferPos = nInitialBufferPos; nBufferPos < nTimes; ++nBufferPos ) {
int nSamplePos = ( int ) fSamplePos;
double fDiff = fSamplePos - nSamplePos;
if ( ( nSamplePos + 1 ) >= nSampleFrames ) {
//we reach the last audioframe.
//set this last frame to zero do nothin wrong.
fVal_L = 0.0;
fVal_R = 0.0;
} else {
// some interpolation methods need 4 frames data.
float last_l;
float last_r;
if ( ( nSamplePos + 2 ) >= nSampleFrames ) {
last_l = 0.0;
last_r = 0.0;
} else {
last_l = pSample_data_L[nSamplePos + 2];
last_r = pSample_data_R[nSamplePos + 2];
}
switch( __interpolateMode ){
case LINEAR:
fVal_L = pSample_data_L[nSamplePos] * (1 - fDiff ) + pSample_data_L[nSamplePos + 1] * fDiff;
fVal_R = pSample_data_R[nSamplePos] * (1 - fDiff ) + pSample_data_R[nSamplePos + 1] * fDiff;
break;
case COSINE:
fVal_L = cosine_Interpolate( pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], fDiff);
fVal_R = cosine_Interpolate( pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], fDiff);
break;
case THIRD:
fVal_L = third_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = third_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case CUBIC:
fVal_L = cubic_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = cubic_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
case HERMITE:
fVal_L = hermite_Interpolate( pSample_data_L[ nSamplePos -1], pSample_data_L[nSamplePos], pSample_data_L[nSamplePos + 1], last_l, fDiff);
fVal_R = hermite_Interpolate( pSample_data_R[ nSamplePos -1], pSample_data_R[nSamplePos], pSample_data_R[nSamplePos + 1], last_r, fDiff);
break;
}
}
if ( fVal_L > fInstrPeak_L ) {
fInstrPeak_L = fVal_L;
}
if ( fVal_R > fInstrPeak_R ) {
fInstrPeak_R = fVal_R;
}
__main_out_L[nBufferPos] += fVal_L;
__main_out_R[nBufferPos] += fVal_R;
fSamplePos += fStep;
} //for
}
__playback_instrument->set_peak_l( fInstrPeak_L );
__playback_instrument->set_peak_r( fInstrPeak_R );
return true;
}