made quality-settings (internal processing samplerate, interpolation and so on) independent of audio-devices (final output) and added new qualitySettings-structure to mixer - might be still buggy and HQ-mode is currently not working, this is going to be fixed tomorrow, anyways important preparation for new project-export-dialog

git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@957 0778d3d1-df1d-0410-868b-ea421aaaa00d

src/core/audio/audio_alsa.cpp

@@ -43,9 +43,8 @@
 
 
 
-audioALSA::audioALSA( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer ) :
-	audioDevice( _sample_rate, tLimit<ch_cnt_t>(
+audioALSA::audioALSA( bool & _success_ful, mixer * _mixer ) :
+	audioDevice( tLimit<ch_cnt_t>(
 		configManager::inst()->value( "audioalsa", "channels" ).toInt(),
 					DEFAULT_CHANNELS, SURROUND_CHANNELS ),
 								_mixer ),
@@ -70,7 +69,7 @@ audioALSA::audioALSA( const sample_rate_t _sample_rate, bool & _success_ful,
 	snd_pcm_hw_params_malloc( &m_hwParams );
 	snd_pcm_sw_params_malloc( &m_swParams );
 
-	if( ( err = setHWParams( _sample_rate, channels(),
+	if( ( err = setHWParams( channels(),
 					SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
 	{
 		printf( "Setting of hwparams failed: %s\n",
@@ -267,9 +266,7 @@ void audioALSA::run( void )
 
 
 
-int audioALSA::setHWParams( const sample_rate_t _sample_rate,
-						const ch_cnt_t _channels,
-						snd_pcm_access_t _access )
+int audioALSA::setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access )
 {
 	int err, dir;
 
@@ -320,32 +317,20 @@ int audioALSA::setHWParams( const sample_rate_t _sample_rate,
 
 	// set the sample rate
 	if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
-						_sample_rate, 0 ) ) < 0 )
+						sampleRate(), 0 ) ) < 0 )
 	{
-		int q = 0;
-		if( sampleRate() == SAMPLE_RATES[1] )
-		{
-			q = 1;
-		}
-		if( sampleRate() == 44100 || sampleRate() == 88200 )
-		{
-			SAMPLE_RATES[0] = 48000;
-			SAMPLE_RATES[1] = 96000;
-		}
-		else
-		{
-			SAMPLE_RATES[0] = 44100;
-			SAMPLE_RATES[1] = 88200;
-		}
-		setSampleRate( SAMPLE_RATES[q] );
 		if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
-						SAMPLE_RATES[q], 0 ) ) < 0 )
+				getMixer()->baseSampleRate(), 0 ) ) < 0 )
 		{
 			printf( "Could not set sample rate: %s\n",
 							snd_strerror( err ) );
 			return( err );
 		}
 	}
+	else
+	{
+		setSampleRate( getMixer()->processingSampleRate() );
+	}
 
 	m_periodSize = getMixer()->framesPerPeriod();
 	m_bufferSize = m_periodSize * 8;

src/core/audio/audio_device.cpp

@@ -3,7 +3,7 @@
 /*
  * audio_device.cpp - base-class for audio-devices used by LMMS-mixer
  *
- * Copyright (c) 2004-2007 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -37,20 +37,15 @@
 
 
 
-audioDevice::audioDevice( const sample_rate_t _sample_rate,
-				const ch_cnt_t _channels, mixer * _mixer ) :
-	m_sampleRate( _sample_rate ),
+audioDevice::audioDevice( const ch_cnt_t _channels, mixer * _mixer ) :
+	m_sampleRate( _mixer->processingSampleRate() ),
 	m_channels( _channels ),
 	m_mixer( _mixer ),
 	m_buffer( new surroundSampleFrame[getMixer()->framesPerPeriod()] )
 {
 	int error;
 	if( ( m_srcState = src_new(
-#ifdef HQ_SINC
-					SRC_SINC_BEST_QUALITY,
-#else
-					SRC_SINC_FASTEST,
-#endif
+			getMixer()->qualitySettings().libsrcInterpolation(),
 				SURROUND_CHANNELS, &error ) ) == NULL )
 	{
 		printf( "Error: src_new() failed in audio_device.cpp!\n" );
@@ -102,11 +97,12 @@ fpp_t audioDevice::getNextBuffer( surroundSampleFrame * _ab )
 	lock();
 
 	// now were safe to access the device
-	if( getMixer()->sampleRate() != m_sampleRate )
+	if( getMixer()->processingSampleRate() != m_sampleRate )
 	{
-		resample( b, frames, _ab, getMixer()->sampleRate(),
+		resample( b, frames, _ab, getMixer()->processingSampleRate(),
 								m_sampleRate );
-		frames = frames * m_sampleRate / getMixer()->sampleRate();
+		frames = frames * m_sampleRate /
+					getMixer()->processingSampleRate();
 	}
 	else
 	{
@@ -156,7 +152,7 @@ void audioDevice::renamePort( audioPort * )
 
 
 
-void FASTCALL audioDevice::resample( const surroundSampleFrame * _src,
+void audioDevice::resample( const surroundSampleFrame * _src,
 						const fpp_t _frames,
 						surroundSampleFrame * _dst,
 						const sample_rate_t _src_sr,
@@ -183,7 +179,7 @@ void FASTCALL audioDevice::resample( const surroundSampleFrame * _src,
 
 
 
-Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
+Uint32 audioDevice::convertToS16( const surroundSampleFrame * _ab,
 						const fpp_t _frames,
 						const float _master_gain,
 						int_sample_t * _output_buffer,
@@ -228,8 +224,7 @@ Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
 
 
 
-void FASTCALL audioDevice::clearS16Buffer( int_sample_t * _outbuf,
-							const fpp_t _frames )
+void audioDevice::clearS16Buffer( int_sample_t * _outbuf, const fpp_t _frames )
 {
 #ifdef LMMS_DEBUG
 	assert( _outbuf != NULL );

src/core/audio/audio_file_device.cpp

@@ -40,13 +40,15 @@ audioFileDevice::audioFileDevice( const sample_rate_t _sample_rate,
 					const bitrate_t _min_bitrate,
 					const bitrate_t _max_bitrate,
 					mixer * _mixer ) :
-	audioDevice( _sample_rate, _channels, _mixer ),
+	audioDevice( _channels, _mixer ),
 	m_outputFile( _file ),
 	m_useVbr( _use_vbr ),
 	m_nomBitrate( _nom_bitrate ),
 	m_minBitrate( _min_bitrate ),
 	m_maxBitrate( _max_bitrate )
 {
+	setSampleRate( _sample_rate );
+
 	if( m_outputFile.open( QFile::WriteOnly | QFile::Truncate ) == FALSE )
 	{
 		QMessageBox::critical( NULL,

src/core/audio/audio_jack.cpp

@@ -48,9 +48,8 @@
 
 
 
-audioJACK::audioJACK( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer ) :
-	audioDevice( _sample_rate, tLimit<int>( configManager::inst()->value(
+audioJACK::audioJACK( bool & _success_ful, mixer * _mixer ) :
+	audioDevice( tLimit<int>( configManager::inst()->value(
 					"audiojack", "channels" ).toInt(),
 					DEFAULT_CHANNELS, SURROUND_CHANNELS ),
 								_mixer ),
@@ -114,9 +113,7 @@ audioJACK::audioJACK( const sample_rate_t _sample_rate, bool & _success_ful,
 
 	if( jack_get_sample_rate( m_client ) != sampleRate() )
 	{
-		SAMPLE_RATES[0] = jack_get_sample_rate( m_client );
-		SAMPLE_RATES[1] = 2 * SAMPLE_RATES[0];
-		setSampleRate( SAMPLE_RATES[0] );
+		setSampleRate( jack_get_sample_rate( m_client ) );
 	}
 
 	for( Uint8 ch = 0; ch < channels(); ++ch )

src/core/audio/audio_oss.cpp

@@ -79,9 +79,8 @@
 
 
 
-audioOSS::audioOSS( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer ) :
-	audioDevice( _sample_rate, tLimit<ch_cnt_t>(
+audioOSS::audioOSS( bool & _success_ful, mixer * _mixer ) :
+	audioDevice( tLimit<ch_cnt_t>(
 		configManager::inst()->value( "audiooss", "channels" ).toInt(),
 					DEFAULT_CHANNELS, SURROUND_CHANNELS ),
 								_mixer ),
@@ -180,31 +179,14 @@ audioOSS::audioOSS( const sample_rate_t _sample_rate, bool & _success_ful,
 	}
 	if( value != sampleRate() )
 	{
-		//printf( "Soundcard uses different sample-rate than LMMS "
-		//						"does!\n" );
-		int q = 0;
-		if( sampleRate() == SAMPLE_RATES[1] )
-		{
-			q = 1;
-		}
-		if( sampleRate() == 44100 || sampleRate() == 88200 )
-		{
-			SAMPLE_RATES[0] = 48000;
-			SAMPLE_RATES[1] = 96000;
-		}
-		else
-		{
-			SAMPLE_RATES[0] = 44100;
-			SAMPLE_RATES[1] = 88200;
-		}
-		setSampleRate( SAMPLE_RATES[q] );
-		value = sampleRate();
+		value = getMixer()->baseSampleRate();
 		if ( ioctl( m_audioFD, SNDCTL_DSP_SPEED, &value ) < 0 )
 		{
 			perror( "SNDCTL_DSP_SPEED" );
 			printf( "Couldn't set audio frequency\n" );
 			return;
 		}
+		setSampleRate( value );
 	}
 
 	_success_ful = TRUE;

src/core/audio/audio_sample_recorder.cpp

@@ -5,7 +5,7 @@
  *                             surround-audio-buffers into RAM, maybe later
  *                             also harddisk
  *
- * Copyright (c) 2004-2007 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -34,11 +34,10 @@
 
 
 
-audioSampleRecorder::audioSampleRecorder( const sample_rate_t _sample_rate,
-						const ch_cnt_t _channels,
+audioSampleRecorder::audioSampleRecorder( const ch_cnt_t _channels,
 							bool & _success_ful,
 							mixer * _mixer ) :
-	audioDevice( _sample_rate, _channels, _mixer ),
+	audioDevice( _channels, _mixer ),
 	m_buffers()
 {
 	_success_ful = TRUE;

src/core/audio/audio_sdl.cpp

@@ -3,7 +3,7 @@
 /*
  * audio_sdl.cpp - device-class that performs PCM-output via SDL
  *
- * Copyright (c) 2004-2007 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2004-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -41,9 +41,8 @@
 
 
 
-audioSDL::audioSDL( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer ) :
-	audioDevice( _sample_rate, DEFAULT_CHANNELS, _mixer ),
+audioSDL::audioSDL( bool & _success_ful, mixer * _mixer ) :
+	audioDevice( DEFAULT_CHANNELS, _mixer ),
 	m_outBuf( new surroundSampleFrame[getMixer()->framesPerPeriod()] ),
 	m_convertedBuf_pos( 0 ),
 	m_convertEndian( FALSE ),
@@ -62,7 +61,7 @@ audioSDL::audioSDL( const sample_rate_t _sample_rate, bool & _success_ful,
 		return;
 	}
 
-	m_audioHandle.freq = _sample_rate;
+	m_audioHandle.freq = sampleRate();
 	m_audioHandle.format = AUDIO_S16SYS;	// we want it in byte-order
 						// of system, so we don't have
 						// to convert the buffers

src/core/bb_track_container.cpp

@@ -170,7 +170,7 @@ void bbTrackContainer::updateBBTrack( trackContentObject * _tco )
 
 void bbTrackContainer::play( void )
 {
-	if( engine::getSong()->playing() )
+	if( engine::getSong()->isPlaying() )
 	{
 		if( engine::getSong()->playMode() != song::Mode_PlayBB )
 		{
@@ -182,7 +182,7 @@ void bbTrackContainer::play( void )
 			engine::getSong()->pause();
 		}
 	}
-	else if( engine::getSong()->paused() )
+	else if( engine::getSong()->isPaused() )
 	{
 		engine::getSong()->resumeFromPause();
 	}

src/core/controller.cpp

@@ -1,8 +1,8 @@
 #ifndef SINGLE_SOURCE_COMPILE
 
 /*
- * controller.cpp - implementation of class controller which handles remote-control
- *                  of automatableModels
+ * controller.cpp - implementation of class controller which handles
+ *                  remote-control of automatableModels
  *
  * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
  *
@@ -82,8 +82,7 @@ unsigned int controller::runningFrames()
 // Get position in seconds
 float controller::runningTime()
 {
-	return s_frames /
-			engine::getMixer()->sampleRate() ;
+	return s_frames / engine::getMixer()->processingSampleRate();
 }
 
 

src/core/engine.cpp

@@ -155,7 +155,7 @@ void engine::destroy( void )
 
 void engine::updateFramesPerTick( void )
 {
-	s_framesPerTick = s_mixer->sampleRate() * 60.0f * 4 /
+	s_framesPerTick = s_mixer->processingSampleRate() * 60.0f * 4 /
 				DefaultTicksPerTact / s_song->getTempo();
 }
 

src/core/envelope_and_lfo_parameters.cpp

@@ -380,7 +380,7 @@ void envelopeAndLFOParameters::updateSampleVars( void )
 	engine::getMixer()->lock();
 
 	const float frames_per_env_seg = SECS_PER_ENV_SEGMENT *
-					engine::getMixer()->sampleRate();
+				engine::getMixer()->processingSampleRate();
 	// TODO: Remove the expKnobVals, time should be linear
 	const f_cnt_t predelay_frames = static_cast<f_cnt_t>(
 							frames_per_env_seg *
@@ -464,7 +464,7 @@ void envelopeAndLFOParameters::updateSampleVars( void )
 
 
 	const float frames_per_lfo_oscillation = SECS_PER_LFO_OSCILLATION *
-					engine::getMixer()->sampleRate();
+				engine::getMixer()->processingSampleRate();
 	m_lfoPredelayFrames = static_cast<f_cnt_t>( frames_per_lfo_oscillation *
 				expKnobVal( m_lfoPredelayModel.value() ) );
 	m_lfoAttackFrames = static_cast<f_cnt_t>( frames_per_lfo_oscillation *

src/core/instrument_functions.cpp

@@ -364,7 +364,7 @@ void arpeggiator::processNote( notePlayHandle * _n )
 
 	// number of frames that every note should be played
 	const f_cnt_t arp_frames = (f_cnt_t)( m_arpTimeModel.value() / 1000.0f *
-					engine::getMixer()->sampleRate() );
+				engine::getMixer()->processingSampleRate() );
 	const f_cnt_t gated_frames = (f_cnt_t)( m_arpGateModel.value() *
 							arp_frames / 100.0f );
 

src/core/instrument_sound_shaping.cpp

@@ -162,7 +162,7 @@ void instrumentSoundShaping::processAudioBuffer( sampleFrame * _ab,
 		if( _n->m_filter == NULL )
 		{
 			_n->m_filter = new basicFilters<>(
-					engine::getMixer()->sampleRate() );
+				engine::getMixer()->processingSampleRate() );
 		}
 		_n->m_filter->setFilterType( m_filterModel.value() );
 

src/core/lfo_controller.cpp

@@ -1,8 +1,8 @@
 #ifndef SINGLE_SOURCE_COMPILE
 
 /*
- * lfo_controller.cpp - implementation of class controller which handles remote-control
- *                  of automatableModels
+ * lfo_controller.cpp - implementation of class controller which handles
+ *                      remote-control of automatableModels
  *
  * Copyright (c) 2008 Paul Giblock <drfaygo/at/gmail.com>
  *
@@ -79,7 +79,8 @@ float lfoController::value( int _offset )
 
 		// The new duration in frames 
 		// (Samples/Second) / (periods/second) = (Samples/cycle)
-		int newDuration = engine::getMixer()->sampleRate() / m_lfoSpeedModel.value();
+		int newDuration = engine::getMixer()->processingSampleRate() /
+							m_lfoSpeedModel.value();
 		
 		m_phaseOffset = m_lfoPhaseModel.value() * newDuration / 360.0;
 
@@ -117,8 +118,9 @@ float lfoController::value( int _offset )
 
 	// 44100 frames/sec
 	return m_lfoBaseModel.value() + ( m_lfoAmountModel.value() * 
-			sinf( TWO_PI * float( ( frame+m_phaseOffset ) * m_lfoSpeedModel.value() ) /
-			engine::getMixer()->sampleRate() ) / 2.0f );
+			sinf( TWO_PI * float( ( frame+m_phaseOffset ) *
+						m_lfoSpeedModel.value() ) /
+			engine::getMixer()->processingSampleRate() ) / 2.0f );
 }
 
 

src/core/mixer.cpp

@@ -65,8 +65,6 @@
 
 
 
-sample_rate_t SAMPLE_RATES[QUALITY_LEVELS] = { 44100, 44100*2 } ;
-
 
 
 #define ALIGN_SIZE 64
@@ -255,8 +253,12 @@ mixer::mixer( void ) :
 	m_multiThreaded( QThread::idealThreadCount() > 1 ),
 	m_workers(),
 	m_numWorkers( m_multiThreaded ? QThread::idealThreadCount() : 0 ),
-	m_workerSem( m_numWorkers  ),
-	m_qualityLevel( DEFAULT_QUALITY_LEVEL ),
+	m_workerSem( m_numWorkers ),
+	m_qualitySettings( qualitySettings::Interpolation_Linear,
+				qualitySettings::Oversampling_None,
+				FALSE,	// sample-exact controllers
+				FALSE	// alias-free oscillators
+				),
 	m_masterGain( 1.0f ),
 	m_audioDev( NULL ),
 	m_oldAudioDev( NULL ),
@@ -372,6 +374,37 @@ void mixer::stopProcessing( void )
 
 
 
+sample_rate_t mixer::baseSampleRate( void ) const
+{
+	sample_rate_t sr =
+		configManager::inst()->value( "mixer", "samplerate" ).toInt();
+	if( sr < 44100 )
+	{
+		sr = 44100;
+	}
+	return( sr );
+}
+
+
+
+
+sample_rate_t mixer::outputSampleRate( void ) const
+{
+	return( m_audioDev != NULL ? m_audioDev->sampleRate() :
+							baseSampleRate() );
+}
+
+
+
+
+sample_rate_t mixer::processingSampleRate( void ) const
+{
+	return( outputSampleRate() * m_qualitySettings.sampleRateMultiplier() );
+}
+
+
+
+
 bool mixer::criticalXRuns( void ) const
 {
 	return( ( m_cpuLoad >= 99 &&
@@ -610,8 +643,8 @@ if( COND_NPH )
 	envelopeAndLFOParameters::triggerLFO();
 	controller::triggerFrameCounter();
 
-	const float new_cpu_load = timer.elapsed() / 10000.0f * sampleRate() /
-							m_framesPerPeriod;
+	const float new_cpu_load = timer.elapsed() / 10000.0f *
+				processingSampleRate() / m_framesPerPeriod;
 	m_cpuLoad = tLimit( (int) ( new_cpu_load * 0.1f + m_cpuLoad * 0.9f ), 0,
 									100 );
 
@@ -643,7 +676,7 @@ void mixer::clear( void )
 
 
 
-void FASTCALL mixer::bufferToPort( const sampleFrame * _buf,
+void mixer::bufferToPort( const sampleFrame * _buf,
 					const fpp_t _frames,
 					const f_cnt_t _offset,
 					stereoVolumeVector _vv,
@@ -695,8 +728,7 @@ void FASTCALL mixer::bufferToPort( const sampleFrame * _buf,
 
 
 
-void FASTCALL mixer::clearAudioBuffer( sampleFrame * _ab,
-							const f_cnt_t _frames,
+void mixer::clearAudioBuffer( sampleFrame * _ab, const f_cnt_t _frames,
 							const f_cnt_t _offset )
 {
 	memset( _ab+_offset, 0, sizeof( *_ab ) * _frames );
@@ -705,8 +737,7 @@ void FASTCALL mixer::clearAudioBuffer( sampleFrame * _ab,
 
 
 #ifndef DISABLE_SURROUND
-void FASTCALL mixer::clearAudioBuffer( surroundSampleFrame * _ab,
-							const f_cnt_t _frames,
+void mixer::clearAudioBuffer( surroundSampleFrame * _ab, const f_cnt_t _frames,
 							const f_cnt_t _offset )
 {
 	memset( _ab+_offset, 0, sizeof( *_ab ) * _frames );
@@ -756,25 +787,23 @@ float mixer::peakValueRight( sampleFrame * _ab, const f_cnt_t _frames )
 
 
 
-void mixer::setHighQuality( bool _hq_on )
+void mixer::changeQuality( const struct qualitySettings & _qs )
 {
 	// don't delete the audio-device
 	stopProcessing();
 
-	// set new quality-level...
-	m_qualityLevel = ( _hq_on == TRUE ) ? HIGH_QUALITY_LEVEL :
-							DEFAULT_QUALITY_LEVEL;
+	m_qualitySettings = _qs;
 
 	startProcessing();
 
-	emit( sampleRateChanged() );
-
+	emit sampleRateChanged();
+	emit qualitySettingsChanged();
 }
 
 
 
 
-void FASTCALL mixer::setAudioDevice( audioDevice * _dev, bool _hq )
+void mixer::setAudioDevice( audioDevice * _dev )
 {
 	stopProcessing();
 
@@ -791,7 +820,6 @@ void FASTCALL mixer::setAudioDevice( audioDevice * _dev, bool _hq )
 		m_audioDev = _dev;
 	}
 
-	m_qualityLevel = _hq ? HIGH_QUALITY_LEVEL : DEFAULT_QUALITY_LEVEL;
 	emit sampleRateChanged();
 
 	startProcessing();
@@ -808,17 +836,8 @@ void mixer::restoreAudioDevice( void )
 		delete m_audioDev;
 
 		m_audioDev = m_oldAudioDev;
-		for( Uint8 qli = DEFAULT_QUALITY_LEVEL;
-						qli < QUALITY_LEVELS; ++qli )
-		{
-			if( SAMPLE_RATES[qli] == m_audioDev->sampleRate() )
-			{
-				m_qualityLevel =
-					static_cast<qualityLevels>( qli );
-				emit sampleRateChanged();
-				break;
-			}
-		}
+		emit sampleRateChanged();
+
 		m_oldAudioDev = NULL;
 		startProcessing();
 	}
@@ -849,7 +868,6 @@ void mixer::removePlayHandles( track * _track )
 
 
 
-
 audioDevice * mixer::tryAudioDevices( void )
 {
 	bool success_ful = FALSE;
@@ -859,8 +877,7 @@ audioDevice * mixer::tryAudioDevices( void )
 #ifdef ALSA_SUPPORT
 	if( dev_name == audioALSA::name() || dev_name == "" )
 	{
-		dev = new audioALSA( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
-							success_ful, this );
+		dev = new audioALSA( success_ful, this );
 		if( success_ful )
 		{
 			m_audioDevName = audioALSA::name();
@@ -874,8 +891,7 @@ audioDevice * mixer::tryAudioDevices( void )
 #ifdef OSS_SUPPORT
 	if( dev_name == audioOSS::name() || dev_name == "" )
 	{
-		dev = new audioOSS( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
-							success_ful, this );
+		dev = new audioOSS( success_ful, this );
 		if( success_ful )
 		{
 			m_audioDevName = audioOSS::name();
@@ -889,8 +905,7 @@ audioDevice * mixer::tryAudioDevices( void )
 #ifdef JACK_SUPPORT
 	if( dev_name == audioJACK::name() || dev_name == "" )
 	{
-		dev = new audioJACK( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
-							success_ful, this );
+		dev = new audioJACK( success_ful, this );
 		if( success_ful )
 		{
 			m_audioDevName = audioJACK::name();
@@ -904,8 +919,7 @@ audioDevice * mixer::tryAudioDevices( void )
 #ifdef SDL_AUDIO_SUPPORT
 	if( dev_name == audioSDL::name() || dev_name == "" )
 	{
-		dev = new audioSDL( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
-							success_ful, this );
+		dev = new audioSDL( success_ful, this );
 		if( success_ful )
 		{
 			m_audioDevName = audioSDL::name();
@@ -929,8 +943,7 @@ audioDevice * mixer::tryAudioDevices( void )
 
 	m_audioDevName = audioDummy::name();
 
-	return( new audioDummy( SAMPLE_RATES[m_qualityLevel], success_ful,
-								this ) );
+	return( new audioDummy( success_ful, this ) );
 }
 
 

src/core/oscillator.cpp

@@ -55,7 +55,7 @@ oscillator::oscillator( const intModel & _wave_shape_model,
 void oscillator::update( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
-	if( m_freq >= engine::getMixer()->sampleRate()/2 )
+	if( m_freq >= engine::getMixer()->processingSampleRate() / 2 )
 	{
 		mixer::clearAudioBuffer( _ab, _frames );
 		return;

src/core/sample_buffer.cpp

@@ -88,7 +88,7 @@ sampleBuffer::sampleBuffer( const QString & _audio_file,
 	m_amplification( 1.0f ),
 	m_reversed( FALSE ),
 	m_frequency( BaseFreq ),
-	m_sampleRate( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
+	m_sampleRate( engine::getMixer()->baseSampleRate() )
 {
 #ifdef SDL_SDL_SOUND_H
 	// init sound-file-system of SDL
@@ -117,7 +117,7 @@ sampleBuffer::sampleBuffer( const sampleFrame * _data, const f_cnt_t _frames ) :
 	m_amplification( 1.0f ),
 	m_reversed( FALSE ),
 	m_frequency( BaseFreq ),
-	m_sampleRate( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
+	m_sampleRate( engine::getMixer()->baseSampleRate() )
 {
 	if( _frames > 0 )
 	{
@@ -148,7 +148,7 @@ sampleBuffer::sampleBuffer( const f_cnt_t _frames ) :
 	m_amplification( 1.0f ),
 	m_reversed( FALSE ),
 	m_frequency( BaseFreq ),
-	m_sampleRate( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
+	m_sampleRate( engine::getMixer()->baseSampleRate() )
 {
 	if( _frames > 0 )
 	{
@@ -205,7 +205,7 @@ void sampleBuffer::update( bool _keep_settings )
 		char * f = qstrdup( file.toAscii().constData() );
 		int_sample_t * buf = NULL;
 		ch_cnt_t channels = DEFAULT_CHANNELS;
-		sample_rate_t samplerate = SAMPLE_RATES[DEFAULT_QUALITY_LEVEL];
+		sample_rate_t samplerate = engine::getMixer()->baseSampleRate();
 
 		m_frames = 0;
 
@@ -279,7 +279,7 @@ m_data[frame][chnl] = buf[idx] * fac;
 
 			delete[] buf;
 
-			normalize_sample_rate( samplerate, _keep_settings );
+			normalizeSampleRate( samplerate, _keep_settings );
 		}
 		else
 		{
@@ -314,14 +314,14 @@ m_data[frame][chnl] = buf[idx] * fac;
 
 
 
-void sampleBuffer::normalize_sample_rate( const sample_rate_t _src_sr,
+void sampleBuffer::normalizeSampleRate( const sample_rate_t _src_sr,
 							bool _keep_settings )
 {
 	// do samplerate-conversion to our default-samplerate
-	if( _src_sr != SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] )
+	if( _src_sr != engine::getMixer()->baseSampleRate() )
 	{
 		sampleBuffer * resampled = resample( this, _src_sr,
-					SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] );
+					engine::getMixer()->baseSampleRate() );
 		delete[] m_data;
 		m_frames = resampled->frames();
 		m_data = new sampleFrame[m_frames];
@@ -588,8 +588,8 @@ bool FASTCALL sampleBuffer::play( sampleFrame * _ab, handleState * _state,
 		return( FALSE );
 	}
 
-	const double freq_factor = (double) _freq / (double) m_frequency
-			* m_sampleRate / engine::getMixer()->sampleRate();
+	const double freq_factor = (double) _freq / (double) m_frequency *
+		m_sampleRate / engine::getMixer()->processingSampleRate();
 
 	// calculate how many frames we have in requested pitch
 	const f_cnt_t total_frames_for_current_pitch = static_cast<f_cnt_t>( (

src/core/sample_play_handle.cpp

@@ -161,8 +161,8 @@ bool samplePlayHandle::isFromTrack( const track * _track ) const
 f_cnt_t samplePlayHandle::totalFrames( void ) const
 {
 	return( ( m_sampleBuffer->endFrame() - m_sampleBuffer->startFrame() ) *
-				( engine::getMixer()->sampleRate() /
-					SAMPLE_RATES[DEFAULT_QUALITY_LEVEL] ) );
+			( engine::getMixer()->processingSampleRate() /
+				engine::getMixer()->baseSampleRate() ) );
 }
 
 

src/core/song.cpp

@@ -607,7 +607,7 @@ void song::startExport( void )
 
 
 
-void song::stopExport( void )
+void song::cancelExport( void )
 {
 	stop();
 	m_exporting = FALSE;

src/gui/automation_editor.cpp

@@ -614,7 +614,7 @@ void automationEditor::keyPressEvent( QKeyEvent * _ke )
 			break;
 
 		case Qt::Key_Space:
-			if( engine::getSong()->playing() )
+			if( engine::getSong()->isPlaying() )
 			{
 				stop();
 			}
@@ -1641,12 +1641,12 @@ void automationEditor::play( void )
 			m_playButton->setIcon( embed::getIconPixmap(
 								"pause" ) );
 		}
-		else if( engine::getSong()->playing() )
+		else if( engine::getSong()->isPlaying() )
 		{
 			engine::getSong()->pause();
 			m_playButton->setIcon( embed::getIconPixmap( "play" ) );
 		}
-		else if( engine::getSong()->paused() )
+		else if( engine::getSong()->isPaused() )
 		{
 			engine::getSong()->resumeFromPause();
 			m_playButton->setIcon( embed::getIconPixmap(
@@ -1662,7 +1662,7 @@ void automationEditor::play( void )
 	}
 	else if( inBBEditor() )
 	{
-		if( engine::getSong()->playing() )
+		if( engine::getSong()->isPlaying() )
 		{
 			m_playButton->setIcon( embed::getIconPixmap( "play" ) );
 		}
@@ -1675,12 +1675,12 @@ void automationEditor::play( void )
 	}
 	else
 	{
-		if( engine::getSong()->playing() )
+		if( engine::getSong()->isPlaying() )
 		{
 			engine::getSong()->pause();
 			m_playButton->setIcon( embed::getIconPixmap( "play" ) );
 		}
-		else if( engine::getSong()->paused() )
+		else if( engine::getSong()->isPaused() )
 		{
 			engine::getSong()->resumeFromPause();
 			m_playButton->setIcon( embed::getIconPixmap(
@@ -1979,7 +1979,7 @@ void automationEditor::deleteSelectedValues( void )
 
 void automationEditor::updatePosition( const midiTime & _t )
 {
-	if( ( engine::getSong()->playing() &&
+	if( ( engine::getSong()->isPlaying() &&
 			engine::getSong()->playMode() ==
 					song::Mode_PlayAutomationPattern ) ||
 							m_scrollBack == TRUE )

src/gui/bb_editor.cpp

@@ -162,7 +162,7 @@ void bbEditor::removeBBView( int _bb )
 
 void bbEditor::play( void )
 {
-	if( engine::getSong()->playing() )
+	if( engine::getSong()->isPlaying() )
 	{
 		if( engine::getSong()->playMode() != song::Mode_PlayBB )
 		{
@@ -178,7 +178,7 @@ void bbEditor::play( void )
 								"play" ) );
 		}
 	}
-	else if( engine::getSong()->paused() )
+	else if( engine::getSong()->isPaused() )
 	{
 		engine::getSong()->resumeFromPause();
 		m_playButton->setIcon( embed::getIconPixmap( "pause" ) );
@@ -217,7 +217,7 @@ void bbEditor::keyPressEvent( QKeyEvent * _ke )
 {
 	if ( _ke->key() == Qt::Key_Space )
 	{
-		if( engine::getSong()->playing() )
+		if( engine::getSong()->isPlaying() )
 		{
 			stop();
 		}

src/gui/export_project_dialog.cpp

@@ -242,7 +242,7 @@ void exportProjectDialog::keyPressEvent( QKeyEvent * _ke )
 {
 	if( _ke->key() == Qt::Key_Escape )
 	{
-		if( engine::getSong()->exporting() == FALSE )
+		if( engine::getSong()->isExporting() == FALSE )
 		{
 			accept();
 		}
@@ -258,7 +258,7 @@ void exportProjectDialog::keyPressEvent( QKeyEvent * _ke )
 
 void exportProjectDialog::closeEvent( QCloseEvent * _ce )
 {
-	if( engine::getSong()->exporting() == TRUE )
+	if( engine::getSong()->isExporting() == TRUE )
 	{
 		abortProjectExport();
 		_ce->ignore();
@@ -308,7 +308,7 @@ void exportProjectDialog::exportBtnClicked( void )
 
 	bool success_ful = FALSE;
 	audioFileDevice * dev = fileEncodeDevices[idx].m_getDevInst(
-							DEFAULT_SAMPLE_RATE,
+							44100,
 							DEFAULT_CHANNELS,
 							success_ful,
 							m_fileName,
@@ -357,7 +357,7 @@ void exportProjectDialog::exportBtnClicked( void )
 
 
 
-	engine::getMixer()->setAudioDevice( dev, m_hqmCb->model()->value() );
+	engine::getMixer()->setAudioDevice( dev );//, m_hqmCb->model()->value() );
 	engine::getSong()->startExport();
 
 	delete m_hqmCb;
@@ -365,8 +365,8 @@ void exportProjectDialog::exportBtnClicked( void )
 	song::playPos & pp = engine::getSong()->getPlayPos(
 							song::Mode_PlaySong );
 
-	while( engine::getSong()->exportDone() == FALSE &&
-				engine::getSong()->exporting() == TRUE
+	while( engine::getSong()->isExportDone() == FALSE &&
+				engine::getSong()->isExporting() == TRUE
 							&& !m_deleteFile )
 	{
 		dev->processNextBuffer();
@@ -392,7 +392,7 @@ void exportProjectDialog::exportBtnClicked( void )
 void exportProjectDialog::cancelBtnClicked( void )
 {
 	// is song-export-thread active?
-	if( engine::getSong()->exporting() == TRUE )
+	if( engine::getSong()->isExporting() == TRUE )
 	{
 		// then dispose abort of export
 		abortProjectExport();
@@ -429,7 +429,7 @@ void exportProjectDialog::finishProjectExport( void )
 		engine::getMainWindow()->resetWindowTitle(); 
 	}
 
-	engine::getSong()->stopExport();
+	engine::getSong()->cancelExport();
 
 	// if we rendered file from command line, quit after export
 	if( file_to_render != "" )

src/gui/file_browser.cpp

@@ -542,8 +542,8 @@ void listView::mouseReleaseEvent( QMouseEvent * _me )
 			samplePlayHandle * s = dynamic_cast<samplePlayHandle *>(
 							m_previewPlayHandle );
 			if( s->totalFrames() - s->framesDone() <=
-				static_cast<f_cnt_t>(
-					engine::getMixer()->sampleRate() * 3 ) )
+				static_cast<f_cnt_t>( engine::getMixer()->
+						processingSampleRate() * 3 ) )
 			{
 				s->setDoneMayReturnTrue( TRUE );
 				m_previewPlayHandle = NULL;

src/gui/piano_roll.cpp

@@ -779,7 +779,7 @@ void pianoRoll::keyPressEvent( QKeyEvent * _ke )
 			break;
 
 		case Qt::Key_Space:
-			if( engine::getSong()->playing() )
+			if( engine::getSong()->isPlaying() )
 			{
 				stop();
 			}
@@ -1087,7 +1087,7 @@ void pianoRoll::mousePressEvent( QMouseEvent * _me )
 
 		// was there an action where should be played the note?
 		if( play_note == TRUE && m_recording == FALSE &&
-					engine::getSong()->playing() == FALSE )
+					engine::getSong()->isPlaying() == FALSE )
 		{
 			m_lastKey = key_num;
 			m_pattern->getInstrumentTrack()->processInEvent(
@@ -1165,7 +1165,7 @@ void pianoRoll::mouseMoveEvent( QMouseEvent * _me )
 				m_action != SELECT_NOTES &&
 				m_action != MOVE_SELECTION &&
 				m_recording == FALSE &&
-				engine::getSong()->playing() == FALSE )
+				engine::getSong()->isPlaying() == FALSE )
 			{
 				m_lastKey = key_num;
 				m_pattern->getInstrumentTrack()->processInEvent(
@@ -1217,7 +1217,7 @@ void pianoRoll::mouseMoveEvent( QMouseEvent * _me )
 			}
 
 			if( shortNote != m_currentNote && 
-					engine::getSong()->playing() == FALSE )
+					engine::getSong()->isPlaying() == FALSE )
 			{
 				if( m_currentNote != NULL ) {
 					m_pattern->getInstrumentTrack()->processInEvent(
@@ -2139,7 +2139,7 @@ void pianoRoll::play( void )
 		return;
 	}
 
-	if( engine::getSong()->playing() )
+	if( engine::getSong()->isPlaying() )
 	{
 		if( engine::getSong()->playMode() != song::Mode_PlayPattern )
 		{
@@ -2154,7 +2154,7 @@ void pianoRoll::play( void )
 			m_playButton->setIcon( embed::getIconPixmap( "play" ) );
 		}
 	}
-	else if( engine::getSong()->paused() )
+	else if( engine::getSong()->isPaused() )
 	{
 		engine::getSong()->resumeFromPause();
 		m_playButton->setIcon( embed::getIconPixmap( "pause" ) );
@@ -2171,7 +2171,7 @@ void pianoRoll::play( void )
 
 void pianoRoll::record( void )
 {
-	if( engine::getSong()->playing() )
+	if( engine::getSong()->isPlaying() )
 	{
 		stop();
 	}
@@ -2526,7 +2526,7 @@ void pianoRoll::deleteSelectedNotes( void )
 
 void pianoRoll::updatePosition( const midiTime & _t )
 {
-	if( ( engine::getSong()->playing() &&
+	if( ( engine::getSong()->isPlaying() &&
 			engine::getSong()->playMode() ==
 					song::Mode_PlayPattern ) ||
 							m_scrollBack == TRUE )

src/gui/setup_dialog.cpp

@@ -748,7 +748,7 @@ void setupDialog::setBufferSize( int _value )
 	m_bufSizeLbl->setText( tr( "Frames: %1\nLatency: %2 ms" ).arg(
 					m_bufferSize ).arg(
 						1000.0f * m_bufferSize /
-					engine::getMixer()->sampleRate(),
+				engine::getMixer()->processingSampleRate(),
 						0, 'f', 1 ) );
 }
 

src/gui/song_editor.cpp

@@ -429,7 +429,7 @@ void songEditor::keyPressEvent( QKeyEvent * _ke )
 	}
 	else if( _ke->key() == Qt::Key_Space )
 	{
-		if( m_s->playing() )
+		if( m_s->isPlaying() )
 		{
 			m_s->stop();
 		}
@@ -598,7 +598,7 @@ void songEditor::masterPitchReleased( void )
 
 void songEditor::updatePosition( const midiTime & _t )
 {
-	if( ( m_s->m_playing && m_s->m_playMode == song::Mode_PlaySong ) ||
+	if( ( m_s->isPlaying() && m_s->m_playMode == song::Mode_PlaySong ) ||
 							m_scrollBack == TRUE )
 	{
 		const int w = width() - DEFAULT_SETTINGS_WIDGET_WIDTH

src/gui/widgets/envelope_and_lfo_view.cpp

@@ -501,7 +501,7 @@ void envelopeAndLFOView::paintEvent( QPaintEvent * )
 	int graph_y_base = LFO_GRAPH_Y + 3 + LFO_GRAPH_H / 2;
 
 	const float frames_for_graph = SECS_PER_LFO_OSCILLATION *
-					engine::getMixer()->sampleRate() / 10;
+				engine::getMixer()->baseSampleRate() / 10;
 
 	const float lfo_gray_amount = 1.0f - fabsf( m_lfoAmountKnob->value() );
 	p.setPen( QPen( QColor( static_cast<int>( 96 * lfo_gray_amount ),

src/tracks/instrument_track.cpp

@@ -551,7 +551,7 @@ bool FASTCALL instrumentTrack::play( const midiTime & _start,
 		{
 			cur_start -= p->startPosition();
 		}
-		if( p->frozen() && engine::getSong()->exporting() == FALSE )
+		if( p->frozen() && !engine::getSong()->isExporting() )
 		{
 			if( cur_start > 0 )
 			{

src/tracks/pattern.cpp

@@ -397,7 +397,7 @@ void pattern::clear( void )
 
 void pattern::freeze( void )
 {
-	if( engine::getSong()->playing() )
+	if( engine::getSong()->isPlaying() )
 	{
 		QMessageBox::information( 0, tr( "Cannot freeze pattern" ),
 						tr( "The pattern currently "
@@ -663,11 +663,9 @@ void patternFreezeThread::run( void )
 	// mixer::restoreAudioDevice(...) deletes old audio-dev and thus
 	// audioSampleRecorder would be destroyed two times...
 	audioSampleRecorder * freeze_recorder = new audioSampleRecorder(
-					engine::getMixer()->sampleRate(),
 							DEFAULT_CHANNELS, b,
 							engine::getMixer() );
-	engine::getMixer()->setAudioDevice( freeze_recorder,
-					engine::getMixer()->highQuality() );
+	engine::getMixer()->setAudioDevice( freeze_recorder );
 
 	// prepare stuff for playing correct things later
 	engine::getSong()->playPattern( m_pattern, FALSE );
@@ -777,7 +775,7 @@ patternView::~patternView()
 		engine::getPianoRoll()->setCurrentPattern( NULL );
 		// we have to have the song-editor to stop playing if it played
 		// us before
-		if( engine::getSong()->playing() &&
+		if( engine::getSong()->isPlaying() &&
 			engine::getSong()->playMode() ==
 							song::Mode_PlayPattern )
 		{