Пример #1
0
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;
}
Пример #2
0
int Sampler::__render_note_no_resample(
	Sample *pSample,
	Note *pNote,
	int nBufferSize,
	int nInitialSilence,
	float cost_L,
	float cost_R,
	float cost_track_L,
	float cost_track_R,
	Song* pSong
)
{
	AudioOutput* audio_output = Hydrogen::get_instance()->getAudioOutput();
	int retValue = 1; // the note is ended

	int nNoteLength = -1;
	if ( pNote->get_length() != -1 ) {
		nNoteLength = ( int )( pNote->get_length() * audio_output->m_transport.m_nTickSize );
	}

	int nAvail_bytes = pSample->get_frames() - ( int )pNote->get_sample_position();	// verifico il numero di frame disponibili ancora da eseguire

	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;
	int nInitialSamplePos = ( int )pNote->get_sample_position();
	int nSamplePos = nInitialSamplePos;
	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;
	float fVal_L;
	float fVal_R;

	/*
	 * 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
			}
		}

		fADSRValue = pNote->get_adsr()->get_value( 1 );
		fVal_L = pSample_data_L[ nSamplePos ] * fADSRValue;
		fVal_R = pSample_data_R[ nSamplePos ] * 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;

		++nSamplePos;
	}
	pNote->update_sample_position( nAvail_bytes );
	pNote->get_instrument()->set_peak_l( fInstrPeak_L );
	pNote->get_instrument()->set_peak_r( fInstrPeak_R );


#ifdef H2CORE_HAVE_LADSPA
		float masterVol =  pSong->get_volume();
	// LADSPA
	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;
			int nSamplePos = nInitialSamplePos;
			for ( int i = 0; i < nAvail_bytes; ++i ) {
				pBuf_L[ nBufferPos ] += pSample_data_L[ nSamplePos ] * fFXCost_L;
				pBuf_R[ nBufferPos ] += pSample_data_R[ nSamplePos ] * fFXCost_R;
				++nSamplePos;
				++nBufferPos;
			}
		}
	}
	// ~LADSPA
#endif

	return retValue;
}
Пример #3
0
bool Sampler::__render_note_no_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,
	Song* pSong
)
{
	AudioOutput* pAudioOutput = Hydrogen::get_instance()->getAudioOutput();
	bool retValue = true; // the note is ended

	int nNoteLength = -1;
	if ( pNote->get_length() != -1 ) {
		nNoteLength = ( int )( pNote->get_length() * pAudioOutput->m_transport.m_nTickSize );
	}

	int nAvail_bytes = pSample->get_frames() - ( int )pSelectedLayerInfo->SamplePosition;	// verifico il numero di frame disponibili ancora da eseguire

	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;
	int nInitialSamplePos = ( int )pSelectedLayerInfo->SamplePosition;
	int nSamplePos = nInitialSamplePos;
	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;
	float fVal_L;
	float fVal_R;


#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 = true;	// the note is ended
			}
		}

		fADSRValue = pNote->get_adsr()->get_value( 1 );
		fVal_L = pSample_data_L[ nSamplePos ] * fADSRValue;
		fVal_R = pSample_data_R[ nSamplePos ] * 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;

		++nSamplePos;
	}
	pSelectedLayerInfo->SamplePosition += nAvail_bytes;
	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 = fLevel * masterVol;
			float fFXCost_R = fLevel * masterVol;

			int nBufferPos = nInitialBufferPos;
			int nSamplePos = nInitialSamplePos;
			for ( int i = 0; i < nAvail_bytes; ++i ) {
				pBuf_L[ nBufferPos ] += pSample_data_L[ nSamplePos ] * fFXCost_L;
				pBuf_R[ nBufferPos ] += pSample_data_R[ nSamplePos ] * fFXCost_R;
				++nSamplePos;
				++nBufferPos;
			}
		}
	}
	// ~LADSPA
#endif

	return retValue;
}
Пример #4
0
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;
}
Пример #5
0
void Mixer::updateMixer()
{
	Preferences *pPref = Preferences::get_instance();
	bool bShowPeaks = pPref->showInstrumentPeaks();

	Hydrogen *pEngine = Hydrogen::get_instance();
	Song *pSong = pEngine->getSong();
	InstrumentList *pInstrList = pSong->get_instrument_list();
	std::vector<DrumkitComponent*>* compoList = pSong->get_components();

	uint nSelectedInstr = pEngine->getSelectedInstrumentNumber();

	float fallOff = pPref->getMixerFalloffSpeed();

	uint nMuteClicked = 0;
	int nInstruments = pInstrList->size();
	int nCompo = compoList->size();
	for ( unsigned nInstr = 0; nInstr < MAX_INSTRUMENTS; ++nInstr ) {

		if ( nInstr >= nInstruments ) {	// unused instrument! let's hide and destroy the mixerline!
			if ( m_pMixerLine[ nInstr ] ) {
				delete m_pMixerLine[ nInstr ];
				m_pMixerLine[ nInstr ] = NULL;

				int newWidth = MIXER_STRIP_WIDTH * ( nInstruments + nCompo );
				if ( m_pFaderPanel->width() != newWidth ) {
					m_pFaderPanel->resize( newWidth, height() );
				}
			}
			continue;
		}
		else {
			if ( m_pMixerLine[ nInstr ] == NULL ) {
				// the mixerline doesn't exists..I'll create a new one!
				m_pMixerLine[ nInstr ] = createMixerLine( nInstr );
				m_pFaderHBox->insertWidget( nInstr, m_pMixerLine[ nInstr ] );

				int newWidth = MIXER_STRIP_WIDTH * ( nInstruments + nCompo );
				if ( m_pFaderPanel->width() != newWidth ) {
					m_pFaderPanel->resize( newWidth, height() );
				}
			}
			MixerLine *pLine = m_pMixerLine[ nInstr ];

			Instrument *pInstr = pInstrList->get( nInstr );
			assert( pInstr );

			float fNewPeak_L = pInstr->get_peak_l();
			pInstr->set_peak_l( 0.0f );	// reset instrument peak

			float fNewPeak_R = pInstr->get_peak_r();
			pInstr->set_peak_r( 0.0f );	// reset instrument peak

			float fNewVolume = pInstr->get_volume();
			bool bMuted = pInstr->is_muted();

			QString sName = pInstr->get_name();
			float fPan_L = pInstr->get_pan_l();
			float fPan_R = pInstr->get_pan_r();


			// fader
			float fOldPeak_L = pLine->getPeak_L();
			float fOldPeak_R = pLine->getPeak_R();

			if (!bShowPeaks) {
				fNewPeak_L = 0.0f;
				fNewPeak_R = 0.0f;
			}

			if ( fNewPeak_L >= fOldPeak_L) {	// LEFT peak
				pLine->setPeak_L( fNewPeak_L );
			}
			else {
				pLine->setPeak_L( fOldPeak_L / fallOff );
			}
			if ( fNewPeak_R >= fOldPeak_R) {	// Right peak
				pLine->setPeak_R( fNewPeak_R );
			}
			else {
				pLine->setPeak_R( fOldPeak_R / fallOff );
			}

			// fader position
			pLine->setVolume( fNewVolume );

			// mute
			if ( bMuted ) {
				nMuteClicked++;
			}
			pLine->setMuteClicked( bMuted );

			// instr name
			pLine->setName( sName );

			// pan
			float fPanValue = 0.0;
			if (fPan_R == 1.0) {
				fPanValue = 1.0 - (fPan_L / 2.0);
			}
			else {
				fPanValue = fPan_R / 2.0;
			}

			pLine->setPan( fPanValue );

			// activity
			if ( pLine->getActivity() > 0 ) {
				pLine->setActivity( m_pMixerLine[ nInstr ]->getActivity() - 30 );
				pLine->setPlayClicked( true );
			}
			else {
				pLine->setPlayClicked( false );
			}

			for (uint nFX = 0; nFX < MAX_FX; nFX++) {
				pLine->setFXLevel( nFX, pInstr->get_fx_level( nFX ) );
			}

			pLine->setSelected( nInstr == nSelectedInstr );

			pLine->updateMixerLine();
		}
	}

	for (std::vector<DrumkitComponent*>::iterator it = compoList->begin() ; it != compoList->end(); ++it) {
		DrumkitComponent* p_compo = *it;

		if( m_pComponentMixerLine.find(p_compo->get_id()) == m_pComponentMixerLine.end() ) {
			// the mixerline doesn't exists..I'll create a new one!
			m_pComponentMixerLine[ p_compo->get_id() ] = createComponentMixerLine( p_compo->get_id() );
			m_pFaderHBox->addWidget( m_pComponentMixerLine[ p_compo->get_id() ] );

			int newWidth = MIXER_STRIP_WIDTH * ( nInstruments + nCompo );
			if ( m_pFaderPanel->width() != newWidth ) {
				m_pFaderPanel->resize( newWidth, height() );
			}
		}

		ComponentMixerLine *pLine = m_pComponentMixerLine[ p_compo->get_id() ];

		float fNewPeak_L = p_compo->get_peak_l();
		p_compo->set_peak_l( 0.0f );	// reset instrument peak

		float fNewPeak_R = p_compo->get_peak_r();
		p_compo->set_peak_r( 0.0f );	// reset instrument peak

		float fNewVolume = p_compo->get_volume();
		bool bMuted = p_compo->is_muted();

		QString sName = p_compo->get_name();

		float fOldPeak_L = pLine->getPeak_L();
		float fOldPeak_R = pLine->getPeak_R();

		if (!bShowPeaks) {
			fNewPeak_L = 0.0f;
			fNewPeak_R = 0.0f;
		}

		if ( fNewPeak_L >= fOldPeak_L) {	// LEFT peak
			pLine->setPeak_L( fNewPeak_L );
		}
		else {
			pLine->setPeak_L( fOldPeak_L / fallOff );
		}
		if ( fNewPeak_R >= fOldPeak_R) {	// Right peak
			pLine->setPeak_R( fNewPeak_R );
		}
		else {
			pLine->setPeak_R( fOldPeak_R / fallOff );
		}

		// fader position
		pLine->setVolume( fNewVolume );

		// mute
		if ( bMuted ) {
			nMuteClicked++;
		}
		pLine->setMuteClicked( bMuted );

		// instr name
		pLine->setName( sName );

		pLine->updateMixerLine();
	}

	if( compoList->size() < m_pComponentMixerLine.size() ) {
		std::vector<int>* p_ids_to_delete = new std::vector<int>();
		for (std::map<int, ComponentMixerLine*>::iterator it=m_pComponentMixerLine.begin(); it!=m_pComponentMixerLine.end(); ++it) {

			bool p_foundExistingRelatedComponent = false;
			for ( std::vector<DrumkitComponent*>::iterator it2 = compoList->begin() ; it2 != compoList->end(); ++it2 ) {
				DrumkitComponent* p_compo = *it2;
				if( p_compo->get_id() == it->first ) {
					p_foundExistingRelatedComponent = true;
					break;
				}
			}
			if( !p_foundExistingRelatedComponent )
				p_ids_to_delete->push_back( it->first ) ;
		}

		for ( std::vector<int>::iterator it = p_ids_to_delete->begin() ; it != p_ids_to_delete->end(); ++it ) {
			int p_compoID = *it;
			delete m_pComponentMixerLine[p_compoID];
			m_pComponentMixerLine.erase( p_compoID );

			int newWidth = MIXER_STRIP_WIDTH * ( nInstruments + nCompo );
			if ( m_pFaderPanel->width() != newWidth ) {
				m_pFaderPanel->resize( newWidth, height() );
			}
		}
	}

	if (nMuteClicked == nInstruments - 1) {
		// find the not muted button
		for (uint i = 0; i < nInstruments; i++) {
			Instrument *instr = pInstrList->get(i);
			if (instr->is_muted() == false) {
				m_pMixerLine[i]->setSoloClicked(true);
				break;
			}
		}
	}
	else {
		for (uint i = 0; i < nInstruments; i++) {
			m_pMixerLine[i]->setSoloClicked(false);
		}
	}


	// update MasterPeak
	float oldPeak_L = m_pMasterLine->getPeak_L();
	float newPeak_L = pEngine->getMasterPeak_L();
	pEngine->setMasterPeak_L(0.0);
	float oldPeak_R = m_pMasterLine->getPeak_R();
	float newPeak_R = pEngine->getMasterPeak_R();
	pEngine->setMasterPeak_R(0.0);

	if (!bShowPeaks) {
		newPeak_L = 0.0;
		newPeak_R = 0.0;
	}

	if (newPeak_L >= oldPeak_L) {
		m_pMasterLine->setPeak_L( newPeak_L );
	}
	else {
		m_pMasterLine->setPeak_L( oldPeak_L / fallOff );
	}
	if (newPeak_R >= oldPeak_R) {
		m_pMasterLine->setPeak_R(newPeak_R);
	}
	else {
		m_pMasterLine->setPeak_R( oldPeak_R / fallOff );
	}




	// set master fader position
	float newVolume = pSong->get_volume();
	float oldVolume = m_pMasterLine->getVolume();
	if (oldVolume != newVolume) {
		m_pMasterLine->setVolume(newVolume);
	}
	m_pMasterLine->updateMixerLine();


#ifdef H2CORE_HAVE_LADSPA
	// LADSPA
	for (uint nFX = 0; nFX < MAX_FX; nFX++) {
		LadspaFX *pFX = Effects::get_instance()->getLadspaFX( nFX );
		if ( pFX ) {
			m_pLadspaFXLine[nFX]->setName( pFX->getPluginName() );
			float fNewPeak_L = 0.0;
			float fNewPeak_R = 0.0;

			float fOldPeak_L = 0.0;
			float fOldPeak_R = 0.0;
			m_pLadspaFXLine[nFX]->getPeaks( &fOldPeak_L, &fOldPeak_R );

			if (fNewPeak_L < fOldPeak_L)	fNewPeak_L = fOldPeak_L / fallOff;
			if (fNewPeak_R < fOldPeak_R)	fNewPeak_R = fOldPeak_R / fallOff;
			m_pLadspaFXLine[nFX]->setPeaks( fNewPeak_L, fNewPeak_R );
			m_pLadspaFXLine[nFX]->setFxActive( pFX->isEnabled() );
			m_pLadspaFXLine[nFX]->setVolume( pFX->getVolume() );
		}
		else {
			m_pLadspaFXLine[nFX]->setName( "No plugin" );
			m_pLadspaFXLine[nFX]->setFxActive( false );
			m_pLadspaFXLine[nFX]->setVolume( 0.0 );
		}
	}
	// ~LADSPA
#endif
}