renamed "mixer::framesPerAudioBuffer()" to "mixer::framesPerPeriod()" and type "fpab_t" to "fpp_t"

git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@502 0778d3d1-df1d-0410-868b-ea421aaaa00d

ChangeLog

@@ -1,3 +1,13 @@
+2007-08-04	Tobias Doerffel	<tobydox/at/users/dot/sourceforge/dot/net>
+
+	* include/mixer.h:
+	* most other files:
+	renamed "framesPerAudioBuffer()" to "framesPerPeriod()"
+
+	* include/types.h:
+	* most other files:
+	renamed type "fpab_t" to "fpp_t"
+
 2007-07-30	Tobias Doerffel	<tobydox/at/users/dot/sourceforge/dot/net>
 
 	* plugins/singerbot/singerbot.h:

include/audio_device.h

@@ -144,29 +144,29 @@ protected:
 	// subclasses can re-implement this for being used in conjunction with
 	// processNextBuffer()
 	virtual void FASTCALL writeBuffer( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const float _master_gain )
 	{
 	}
 
 	// called by according driver for fetching new sound-data
-	fpab_t FASTCALL getNextBuffer( surroundSampleFrame * _ab );
+	fpp_t FASTCALL getNextBuffer( surroundSampleFrame * _ab );
 
 	// convert a given audio-buffer to a buffer in signed 16-bit samples
 	// returns num of bytes in outbuf
 	Uint32 FASTCALL convertToS16( const surroundSampleFrame * _ab,
-					const fpab_t _frames,
+					const fpp_t _frames,
 					const float _master_gain,
 					int_sample_t * _output_buffer,
 					const bool _convert_endian = FALSE );
 
 	// clear given signed-int-16-buffer
 	void FASTCALL clearS16Buffer( int_sample_t * _outbuf,
-							const fpab_t _frames );
+							const fpp_t _frames );
 
 	// resample given buffer from samplerate _src_sr to samplerate _dst_sr
 	void FASTCALL resample( const surroundSampleFrame * _src,
-					const fpab_t _frames,
+					const fpp_t _frames,
 					surroundSampleFrame * _dst,
 					const sample_rate_t _src_sr,
 					const sample_rate_t _dst_sr );

include/audio_dummy.h

@@ -109,7 +109,7 @@ private:
 			delete[] b;
 
 			const Sint32 microseconds = static_cast<Sint32>(
-					getMixer()->framesPerAudioBuffer() *
+					getMixer()->framesPerPeriod() *
 					1000000.0f / getMixer()->sampleRate() -
 							timer.elapsed() );
 			if( microseconds > 0 )

include/audio_file_ogg.h

@@ -72,7 +72,7 @@ public:
 
 private:
 	virtual void FASTCALL writeBuffer( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const float _master_gain );
 
 	bool startEncoding( void );

include/audio_file_wave.h

@@ -66,7 +66,7 @@ public:
 
 private:
 	virtual void FASTCALL writeBuffer( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						float _master_gain );
 
 	bool startEncoding( void );

include/audio_port.h

@@ -109,7 +109,7 @@ private:
 	QString m_name;
 	
 	effectChain * m_effects;
-	fpab_t m_frames;
+	fpp_t m_frames;
 
 } ;
 

include/audio_sample_recorder.h

@@ -64,10 +64,10 @@ public:
 
 private:
 	virtual void FASTCALL writeBuffer( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const float _master_gain );
 
-	typedef vlist<QPair<sampleFrame *, fpab_t> > bufferList;
+	typedef vlist<QPair<sampleFrame *, fpp_t> > bufferList;
 	bufferList m_buffers;
 
 } ;

include/automation_track.h

@@ -47,7 +47,7 @@ private:
 	virtual trackTypes type( void ) const;
 
 	virtual bool FASTCALL play( const midiTime & _start,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const f_cnt_t _frame_base,
 							Sint16 _tco_num = -1 );
 

include/bb_editor.h

@@ -40,7 +40,7 @@ class bbEditor : public trackContainer
 {
 	Q_OBJECT
 public:
-	virtual bool FASTCALL play( midiTime _start, const fpab_t _frames,
+	virtual bool FASTCALL play( midiTime _start, const fpp_t _frames,
 						const f_cnt_t _frame_base,
 							Sint16 _tco_num = -1 );
 

include/bb_track.h

@@ -102,7 +102,7 @@ public:
 
 	virtual trackTypes type( void ) const;
 	virtual bool FASTCALL play( const midiTime & _start,
-					const fpab_t _frames,
+					const fpp_t _frames,
 					const f_cnt_t _frame_base,
 							Sint16 _tco_num = -1 );
 	virtual trackContentObject * FASTCALL createTCO( const midiTime &

include/effect.h

@@ -58,7 +58,7 @@ public:
 	virtual ~effect();
 	
 	virtual bool FASTCALL processAudioBuffer( 
-			surroundSampleFrame * _buf, const fpab_t _frames );
+			surroundSampleFrame * _buf, const fpp_t _frames );
 	
 	inline ch_cnt_t getProcessorCount( void )
 	{

include/effect_chain.h

@@ -42,7 +42,7 @@ public:
 	void FASTCALL moveDown( effect * _effect );
 	void FASTCALL moveUp( effect * _effect );
 	bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf, 
-							const fpab_t _frames );
+							const fpp_t _frames );
 	void startRunning( void );
 	bool isRunning( void );
 	

include/envelope_and_lfo_widget.h

@@ -87,7 +87,7 @@ public:
 
 	void FASTCALL fillLevel( float * _buf, f_cnt_t _frame,
 				const f_cnt_t _release_begin,
-				const fpab_t _frames );
+				const fpp_t _frames );
 
 	inline bool used( void ) const
 	{
@@ -115,7 +115,7 @@ protected:
 	virtual void paintEvent( QPaintEvent * _pe );
 
 	void FASTCALL fillLFOLevel( float * _buf, f_cnt_t _frame,
-							const fpab_t _frames );
+							const fpp_t _frames );
 
 
 protected slots:
@@ -182,7 +182,7 @@ private:
 		USER
 	} m_lfoShape;
 
-	sample_t lfoShapeSample( fpab_t _frame_offset );
+	sample_t lfoShapeSample( fpp_t _frame_offset );
 	void updateLFOShapeData( void );
 
 

include/envelope_tab_widget.h

@@ -64,7 +64,7 @@ public:
 	virtual ~envelopeTabWidget();
 
 	void FASTCALL processAudioBuffer( sampleFrame * _ab,
-							const fpab_t _frames,
+							const fpp_t _frames,
 							notePlayHandle * _n );
 
 	enum targets

include/instrument_track.h

@@ -81,7 +81,7 @@ public:
 
 	// used by instrument
 	void FASTCALL processAudioBuffer( sampleFrame * _buf,
-							const fpab_t _frames,
+							const fpp_t _frames,
 							notePlayHandle * _n );
 
 	virtual void FASTCALL processInEvent( const midiEvent & _me,
@@ -134,7 +134,7 @@ public:
 
 	// play everything in given frame-range - creates note-play-handles
 	virtual bool FASTCALL play( const midiTime & _start,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const f_cnt_t _frame_base,
 							Sint16 _tco_num = -1 );
 	// create new track-content-object = pattern

include/mixer.h

@@ -59,7 +59,7 @@ class midiClient;
 class audioPort;
 
 
-const fpab_t DEFAULT_BUFFER_SIZE = 512;
+const fpp_t DEFAULT_BUFFER_SIZE = 512;
 
 const ch_cnt_t DEFAULT_CHANNELS = 2;
 
@@ -196,9 +196,9 @@ public:
 
 
 	// methods providing information for other classes
-	inline fpab_t framesPerAudioBuffer( void ) const
+	inline fpp_t framesPerPeriod( void ) const
 	{
-		return( m_framesPerAudioBuffer );
+		return( m_framesPerPeriod );
 	}
 
 	inline const surroundSampleFrame * currentAudioBuffer( void ) const
@@ -282,16 +282,18 @@ public:
 
 	// audio-buffer-mgm
 	void FASTCALL bufferToPort( const sampleFrame * _buf,
-					const fpab_t _frames,
-					const f_cnt_t _framesAhead,
+					const fpp_t _frames,
+					const f_cnt_t _offset,
 					const volumeVector & _volumeVector,
 							audioPort * _port );
 
 	void FASTCALL clearAudioBuffer( sampleFrame * _ab,
-							const f_cnt_t _frames );
+						const f_cnt_t _frames,
+						const f_cnt_t _offset = 0 );
 #ifndef DISABLE_SURROUND
 	void FASTCALL clearAudioBuffer( surroundSampleFrame * _ab,
-							const f_cnt_t _frames );
+						const f_cnt_t _frames,
+						const f_cnt_t _offset = 0 );
 #endif
 
 	bool criticalXRuns( void ) const;
@@ -346,13 +348,13 @@ private:
 	void processBuffer( const surroundSampleFrame * _buf,
 						const fx_ch_t _fx_chnl );
 
-	void FASTCALL scaleClip( fpab_t _frame, ch_cnt_t _chnl );
+	void FASTCALL scaleClip( fpp_t _frame, ch_cnt_t _chnl );
 
 	const surroundSampleFrame * renderNextBuffer( void );
 
 	vvector<audioPort *> m_audioPorts;
 
-	fpab_t m_framesPerAudioBuffer;
+	fpp_t m_framesPerPeriod;
 
 	surroundSampleFrame * m_readBuf;
 	surroundSampleFrame * m_writeBuf;
@@ -366,7 +368,7 @@ private:
 	bool m_scaleClip;
 	surroundSampleFrame m_maxClip;
 	surroundSampleFrame m_previousSample;
-	fpab_t m_halfStart[SURROUND_CHANNELS];
+	fpp_t m_halfStart[SURROUND_CHANNELS];
 	bool m_clipped[SURROUND_CHANNELS];
 	bool m_oldBuffer[SURROUND_CHANNELS];
 	bool m_newBuffer[SURROUND_CHANNELS];

include/oscillator.h

@@ -77,7 +77,7 @@ public:
 		m_userWave = _wave;
 	}
 
-	void update( sampleFrame * _ab, const fpab_t _frames,
+	void update( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 
 /*#define	FLOAT_TO_INT(in,out)		\
@@ -166,40 +166,40 @@ private:
 	const sampleBuffer * m_userWave;
 
 
-	void updateNoSub( sampleFrame * _ab, const fpab_t _frames,
+	void updateNoSub( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
-	void updatePM( sampleFrame * _ab, const fpab_t _frames,
+	void updatePM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
-	void updateAM( sampleFrame * _ab, const fpab_t _frames,
+	void updateAM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
-	void updateMix( sampleFrame * _ab, const fpab_t _frames,
+	void updateMix( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
-	void updateSync( sampleFrame * _ab, const fpab_t _frames,
+	void updateSync( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
-	void updateFM( sampleFrame * _ab, const fpab_t _frames,
+	void updateFM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 
-	float syncInit( sampleFrame * _ab, const fpab_t _frames,
+	float syncInit( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 	inline bool syncOk( float _osc_coeff );
 
 	template<waveShapes W>
-	void updateNoSub( sampleFrame * _ab, const fpab_t _frames,
+	void updateNoSub( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 	template<waveShapes W>
-	void updatePM( sampleFrame * _ab, const fpab_t _frames,
+	void updatePM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 	template<waveShapes W>
-	void updateAM( sampleFrame * _ab, const fpab_t _frames,
+	void updateAM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 	template<waveShapes W>
-	void updateMix( sampleFrame * _ab, const fpab_t _frames,
+	void updateMix( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 	template<waveShapes W>
-	void updateSync( sampleFrame * _ab, const fpab_t _frames,
+	void updateSync( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 	template<waveShapes W>
-	void updateFM( sampleFrame * _ab, const fpab_t _frames,
+	void updateFM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl );
 
 	template<waveShapes W>

include/sample_buffer.h

@@ -98,7 +98,7 @@ public:
 	virtual ~sampleBuffer();
 
 	bool FASTCALL play( sampleFrame * _ab, handleState * _state,
-				const fpab_t _frames,
+				const fpp_t _frames,
 				const float _freq = BASE_FREQ,
 				const bool _looped = FALSE );
 

include/sample_play_handle.h

@@ -49,7 +49,6 @@ public:
 	virtual ~samplePlayHandle();
 
 	virtual void play( bool _try_parallelizing );
-	void play( const fpab_t _frame_base, bool _try_parallelizing );
 	virtual bool done( void ) const;
 
 	virtual bool isFromTrack( const track * _track ) const;

include/sample_track.h

@@ -130,7 +130,7 @@ public:
 
 	virtual trackTypes type( void ) const;
 	virtual bool FASTCALL play( const midiTime & _start,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const f_cnt_t _frame_base,
 							Sint16 _tco_num = -1 );
 	virtual trackContentObject * FASTCALL createTCO( const midiTime &

include/sweep_oscillator.h

@@ -48,12 +48,12 @@ public:
 	{
 	}
 
-	void update( sampleFrame * _ab, const fpab_t _frames, 
+	void update( sampleFrame * _ab, const fpp_t _frames, 
 					const float _freq1, const float _freq2,
 						const float _sample_rate )
 	{
 		const float df = _freq2 - _freq1;
-		for( fpab_t frame = 0; frame < _frames; ++frame )
+		for( fpp_t frame = 0; frame < _frames; ++frame )
 		{
 			sample_t s = oscillator::sinSample( m_phase );
 			for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )

include/track.h

@@ -431,7 +431,7 @@ public:
 	virtual trackTypes type( void ) const = 0;
 
 	virtual bool FASTCALL play( const midiTime & _start,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const f_cnt_t _frame_base,
 						Sint16 _tco_num = -1 ) = 0;
 

include/types.h

@@ -47,7 +47,7 @@ typedef Sint16 int_sample_t;		// 16-bit-int-sample
 
 
 typedef Uint32 sample_rate_t;		// sample-rate
-typedef Sint16 fpab_t;			// frames per audio-buffer (0-16384)
+typedef Sint16 fpp_t;			// frames per period (0-16384)
 typedef Sint32 f_cnt_t;			// standard frame-count
 typedef Uint8 ch_cnt_t;			// channel-count (0-SURROUND_CHANNELS)
 typedef Uint16 bpm_t;			// tempo (MIN_BPM to MAX_BPM)

plugins/bass_booster/bass_booster.cpp

@@ -69,7 +69,7 @@ bassBoosterEffect::~bassBoosterEffect()
 
 
 bool FASTCALL bassBoosterEffect::processAudioBuffer( surroundSampleFrame * _buf,
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	if( isBypassed() || !isRunning () )
 	{
@@ -77,7 +77,7 @@ bool FASTCALL bassBoosterEffect::processAudioBuffer( surroundSampleFrame * _buf,
 	}
 
 	double out_sum = 0.0;
-	for( fpab_t f = 0; f < _frames; ++f )
+	for( fpp_t f = 0; f < _frames; ++f )
 	{
 		sample_t s[2] = { _buf[f][0], _buf[f][1] };
 		m_bbFX.nextSample( s[0], s[1] );

plugins/bass_booster/bass_booster.h

@@ -41,7 +41,7 @@ public:
 	bassBoosterEffect( const descriptor::subPluginFeatures::key * _key );
 	virtual ~bassBoosterEffect();
 	virtual bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf,
-							const fpab_t _frames );
+							const fpp_t _frames );
 	inline virtual QString nodeName( void ) const
 	{
 		return( "bassboostereffect" );

plugins/bit_invader/bit_invader.cpp

@@ -701,11 +701,11 @@ void bitInvader::playNote( notePlayHandle * _n, bool )
 					engine::getMixer()->sampleRate() );
 	}
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 
 	bSynth * ps = static_cast<bSynth *>( _n->m_pluginData );
-	for( fpab_t frame = 0; frame < frames; ++frame )
+	for( fpp_t frame = 0; frame < frames; ++frame )
 	{
 		const sample_t cur = ps->nextStringSample();
 		for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )

plugins/kicker/kicker.cpp

@@ -192,12 +192,13 @@ void kickerInstrument::playNote( notePlayHandle * _n, bool )
 
 	//const float freq = getInstrumentTrack()->frequency( _n ) / 2;
 	const float fdiff = m_endFreqKnob->value() - m_startFreqKnob->value();
-	const fpab_t frames = _n->released() ?
+/*	const fpp_t frames = _n->released() ?
 		tMax( tMin<f_cnt_t>( desiredReleaseFrames() -
 							_n->releaseFramesDone(),
 			engine::getMixer()->framesPerAudioBuffer() ), 0 )
 		:
-		engine::getMixer()->framesPerAudioBuffer();
+		engine::getMixer()->framesPerAudioBuffer();*/
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	const float f1 = m_startFreqKnob->value() + tfp * fdiff / decfr;
 	const float f2 = m_startFreqKnob->value() + (frames+tfp-1)*fdiff/decfr;
 
@@ -209,7 +210,7 @@ void kickerInstrument::playNote( notePlayHandle * _n, bool )
 
 	if( _n->released() )
 	{
-		for( fpab_t f = 0; f < frames; ++f )
+		for( fpp_t f = 0; f < frames; ++f )
 		{
 			const float fac = 1.0f -
 				(float)( _n->releaseFramesDone()+f ) /

plugins/ladspa_effect/ladspa_effect.cpp

@@ -116,7 +116,7 @@ ladspaEffect::ladspaEffect( const descriptor::subPluginFeatures::key * _key ) :
 		// during cleanup.  It was easier to troubleshoot with the
 		// memory management all taking place in one file.
 				p->buffer = 
-		new LADSPA_Data[engine::getMixer()->framesPerAudioBuffer()];
+		new LADSPA_Data[engine::getMixer()->framesPerPeriod()];
 				
 				if( p->name.toUpper().contains( "IN" ) && 
 					m_ladspa->isPortInput( m_key, port ) )
@@ -334,7 +334,7 @@ ladspaEffect::~ladspaEffect()
 
 
 bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf, 
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	if( !isOkay() || dontRun() || !isRunning() || isBypassed() )
 	{
@@ -353,7 +353,7 @@ bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
 			switch( m_ports[proc][port]->rate )
 			{
 				case CHANNEL_IN:
-					for( fpab_t frame = 0; 
+					for( fpp_t frame = 0; 
 						frame < _frames; frame++ )
 					{
 						m_ports[proc][port]->buffer[frame] = 
@@ -369,7 +369,7 @@ bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
 					// This only supports control rate ports, so the audio rates are
 					// treated as though they were control rate by setting the
 					// port buffer to all the same value.
-					for( fpab_t frame = 0; 
+					for( fpp_t frame = 0; 
 						frame < _frames; frame++ )
 					{
 						m_ports[proc][port]->buffer[frame] = 
@@ -419,7 +419,7 @@ bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
 				case CONTROL_RATE_INPUT:
 					break;
 				case CHANNEL_OUT:
-					for( fpab_t frame = 0; 
+					for( fpp_t frame = 0; 
 						frame < _frames; frame++ )
 					{
 						_buf[frame][channel] = 

plugins/ladspa_effect/ladspa_effect.h

@@ -50,7 +50,7 @@ public:
 	virtual ~ladspaEffect();
 
 	virtual bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf,
-							const fpab_t _frames );
+							const fpp_t _frames );
 	
 	void FASTCALL setControl( Uint16 _control, LADSPA_Data _data );
 
@@ -89,7 +89,7 @@ private:
 	ladspa2LMMS * m_ladspa;
 	Uint16 m_effectChannels;
 	Uint16 m_portCount;
-	fpab_t m_bufferSize;
+	fpp_t m_bufferSize;
 			
 	const LADSPA_Descriptor * m_descriptor;
 	vvector<LADSPA_Handle> m_handles;

plugins/lb302/lb302.cpp

@@ -738,7 +738,7 @@ void lb302Synth::playNote( notePlayHandle * _n, bool )
         }
     }
     
-    const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
+    const fpp_t frames = _n->framesLeftForCurrentPeriod();
     sampleFrame *buf = new sampleFrame[frames];
 
         process(buf, frames); 

plugins/organic/organic.cpp

@@ -359,7 +359,7 @@ void organicInstrument::playNote( notePlayHandle * _n, bool )
 	oscillator * osc_r = static_cast<oscPtr *>( _n->m_pluginData
 								)->oscRight;
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 	
 	osc_l->update( buf, frames, 0 );

plugins/patman/patman.cpp

@@ -213,7 +213,7 @@ QString patmanSynth::nodeName( void ) const
 
 void patmanSynth::playNote( notePlayHandle * _n, bool )
 {
-	const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 
 	if( !_n->m_pluginData )

plugins/plucked_string_synth/plucked_string_synth.cpp

@@ -143,11 +143,11 @@ void pluckedStringSynth::playNote( notePlayHandle * _n, bool )
 					engine::getMixer()->sampleRate() );
 	}
 
-	const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 
 	pluckSynth * ps = static_cast<pluckSynth *>( _n->m_pluginData );
-	for( Uint32 frame = 0; frame < frames; ++frame )
+	for( fpp_t frame = 0; frame < frames; ++frame )
 	{
 		const sample_t cur = ps->nextStringSample();
 		for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )

plugins/polyb302/polyb302.cpp

@@ -551,7 +551,7 @@ void polyb302Synth::playNote( notePlayHandle * _n, bool )
 		}
 	}
 
-	const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 
 	hstate->process( buf, frames, _n->frequency() ); 

plugins/singerbot/singerbot.cpp

@@ -145,7 +145,7 @@ singerBot::~singerBot()
 
 void singerBot::playNote( notePlayHandle * _n, bool )
 {
-	const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 
 	if( !_n->m_pluginData )
 	{
@@ -293,7 +293,7 @@ void singerBot::createWave( notePlayHandle * _n )
 
 
 void singerBot::play( sampleFrame * _ab, handle_data * _hdata,
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	const f_cnt_t offset = _hdata->wave->num_samples()
 						- _hdata->remaining_frames;

plugins/singerbot/singerbot.h

@@ -144,7 +144,7 @@ private:
 
 	void createWave( notePlayHandle * _n );
 	void play( sampleFrame * _ab, handle_data * _hdata,
-							const fpab_t _frames );
+							const fpp_t _frames );
 	void updateWords( void );
 
 } ;

plugins/stk/mallets/mallets.cpp

@@ -94,7 +94,7 @@ mallets::mallets( instrumentTrack * _channel_track ) :
 	m_spread->move( 178, 173 );
 	m_spread->setHintText( tr( "Spread:" ) + " ", "" );
 
-	m_buffer = new sampleFrame[engine::getMixer()->framesPerAudioBuffer()];
+	m_buffer = new sampleFrame[engine::getMixer()->framesPerPeriod()];
 }
 
 
@@ -441,7 +441,7 @@ void mallets::playNote( notePlayHandle * _n, bool )
 		}
 	}
 
-	const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 
 	malletsSynth * ps = static_cast<malletsSynth *>( _n->m_pluginData );
 	sample_t add_scale = 0.0f;
@@ -450,7 +450,7 @@ void mallets::playNote( notePlayHandle * _n, bool )
 		add_scale = 
 		static_cast<sample_t>( m_strike->value() ) * freq * 2.5f;
 	}
-	for( Uint32 frame = 0; frame < frames; ++frame )
+	for( fpp_t frame = 0; frame < frames; ++frame )
 	{
 		const sample_t left = ps->nextSampleLeft() * 
 				( m_scalers[m_presets->value()] + add_scale );

plugins/triple_oscillator/triple_oscillator.cpp

@@ -686,7 +686,7 @@ void tripleOscillator::playNote( notePlayHandle * _n, bool )
 	oscillator * osc_r = static_cast<oscPtr *>( _n->m_pluginData
 								)->oscRight;
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 	
 	osc_l->update( buf, frames, 0 );

plugins/vestige/vestige.cpp

@@ -302,7 +302,7 @@ void vestigeInstrument::waitForWorkerThread( void )
 		return;
 	}
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = engine::getMixer()->framesPerPeriod();
 	sampleFrame * buf = new sampleFrame[frames];
 
 	if( m_plugin->waitForProcessingFinished( buf ) )
@@ -358,7 +358,7 @@ void vestigeInstrument::playNote( notePlayHandle * _n, bool )
 			m_noteStates[k] = ON;
 		}
 		m_plugin->enqueueMidiEvent( midiEvent( NOTE_ON, 0, k,
-					_n->getVolume() ), _n->framesAhead() );
+					_n->getVolume() ), _n->offset() );
 		// notify when the handle stops, call to deleteNotePluginData
 		_n->m_pluginData = _n;
 	}

plugins/vibed/vibed.cpp

@@ -575,13 +575,13 @@ void vibed::playNote( notePlayHandle * _n, bool )
 		}
 	}
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = _n->framesLeftForCurrentPeriod();
 	stringContainer * ps = static_cast<stringContainer *>(
 			 				_n->m_pluginData );
 	
 	sampleFrame * buf = new sampleFrame[frames];
 
-	for( fpab_t i = 0; i < frames; ++i )
+	for( fpp_t i = 0; i < frames; ++i )
 	{
 		buf[i][0] = 0.0f;
 		buf[i][1] = 0.0f;

plugins/vst_base/lvsl_client.cpp

@@ -289,7 +289,7 @@ void remoteVSTPlugin::hideEditor( void )
 bool remoteVSTPlugin::process( const sampleFrame * _in_buf,
 					sampleFrame * _out_buf, bool _wait )
 {
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = engine::getMixer()->framesPerPeriod();
 
 	if( m_shm == NULL )
 	{
@@ -317,7 +317,7 @@ bool remoteVSTPlugin::process( const sampleFrame * _in_buf,
 	{
 		for( ch_cnt_t ch = 0; ch < inputs; ++ch )
 		{
-			for( fpab_t frame = 0; frame < frames; ++frame )
+			for( fpp_t frame = 0; frame < frames; ++frame )
 			{
 				m_shm[ch * frames + frame] = _in_buf[frame][ch];
 			}
@@ -353,7 +353,7 @@ bool remoteVSTPlugin::waitForProcessingFinished( sampleFrame * _out_buf )
 		usleep( 10 );
 	}
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = engine::getMixer()->framesPerPeriod();
 	const ch_cnt_t outputs = tMax<ch_cnt_t>( m_outputCount,
 							DEFAULT_CHANNELS );
 	if( outputs != DEFAULT_CHANNELS )
@@ -364,7 +364,7 @@ bool remoteVSTPlugin::waitForProcessingFinished( sampleFrame * _out_buf )
 
 	for( ch_cnt_t ch = 0; ch < outputs; ++ch )
 	{
-		for( fpab_t frame = 0; frame < frames; ++frame )
+		for( fpp_t frame = 0; frame < frames; ++frame )
 		{
 			_out_buf[frame][ch] = m_shm[(m_inputCount+ch)*
 								frames+frame];
@@ -571,8 +571,8 @@ Sint16 remoteVSTPlugin::processNextMessage( void )
 		case VST_GET_BUFFER_SIZE:
 			writeValueS<Sint16>( VST_BUFFER_SIZE );
 			// handle is the same
-			writeValueS<fpab_t>(
-				engine::getMixer()->framesPerAudioBuffer() );
+			writeValueS<fpp_t>(
+				engine::getMixer()->framesPerPeriod() );
 			break;
 
 		case VST_SHM_KEY_AND_SIZE:

plugins/vst_base/lvsl_server.c

@@ -161,7 +161,7 @@ public:
 	}
 
 	// set given block-size for plugin
-	void setBlockSize( const fpab_t _bsize );
+	void setBlockSize( const fpp_t _bsize );
 
 	// set given tempo
 	void setBPM( const bpm_t _bpm )
@@ -244,7 +244,7 @@ private:
 	DWORD m_guiThreadID;
 
 
-	fpab_t m_blockSize;
+	fpp_t m_blockSize;
 	float * m_shm;
 
 	float * * m_inputs;
@@ -532,7 +532,7 @@ void VSTPlugin::enqueueMidiEvent( const midiEvent & _event,
 
 
 
-void VSTPlugin::setBlockSize( const fpab_t _bsize )
+void VSTPlugin::setBlockSize( const fpp_t _bsize )
 {
 	if( _bsize == m_blockSize )
 	{
@@ -1219,7 +1219,7 @@ int main( void )
 
 
 			case VST_BUFFER_SIZE:
-				plugin->setBlockSize( readValue<fpab_t>() );
+				plugin->setBlockSize( readValue<fpp_t>() );
 				break;
 
 			case VST_BPM:

plugins/vst_effect/vst_effect.cpp

@@ -88,7 +88,7 @@ vstEffect::~vstEffect()
 
 
 bool FASTCALL vstEffect::processAudioBuffer( surroundSampleFrame * _buf, 
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	if( isBypassed() || !isRunning () )
 	{
@@ -98,7 +98,7 @@ bool FASTCALL vstEffect::processAudioBuffer( surroundSampleFrame * _buf,
 	if( m_plugin )
 	{
 		sampleFrame * buf = new sampleFrame[_frames];
-		for( fpab_t f = 0; f < _frames; ++f )
+		for( fpp_t f = 0; f < _frames; ++f )
 		{
 			for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
 			{
@@ -109,7 +109,7 @@ bool FASTCALL vstEffect::processAudioBuffer( surroundSampleFrame * _buf,
 		m_plugin->process( buf, buf, TRUE );
 		m_pluginMutex.unlock();
 		double out_sum = 0.0;
-		for( fpab_t f = 0; f < _frames; ++f )
+		for( fpp_t f = 0; f < _frames; ++f )
 		{
 			for( ch_cnt_t ch = 0; ch < SURROUND_CHANNELS; ++ch )
 			{

plugins/vst_effect/vst_effect.h

@@ -53,7 +53,7 @@ public:
 	virtual ~vstEffect();
 
 	virtual bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf,
-							const fpab_t _frames );
+							const fpp_t _frames );
 
 	virtual inline QString publicName( void ) const
 	{

src/audio/audio_alsa.cpp

@@ -213,13 +213,13 @@ void audioALSA::stopProcessing( void )
 void audioALSA::run( void )
 {
 	surroundSampleFrame * temp =
-		new surroundSampleFrame[getMixer()->framesPerAudioBuffer()];
+		new surroundSampleFrame[getMixer()->framesPerPeriod()];
 	int_sample_t * outbuf =
-			new int_sample_t[getMixer()->framesPerAudioBuffer() *
+			new int_sample_t[getMixer()->framesPerPeriod() *
 								channels()];
 	int_sample_t * pcmbuf = new int_sample_t[m_periodSize * channels()];
 
-	int outbuf_size = getMixer()->framesPerAudioBuffer() * channels();
+	int outbuf_size = getMixer()->framesPerPeriod() * channels();
 	int outbuf_pos = 0;
 	int pcmbuf_size = m_periodSize * channels();
 
@@ -233,7 +233,7 @@ void audioALSA::run( void )
 			if( outbuf_pos == 0 )
 			{
 				// frames depend on the sample rate
-				const fpab_t frames = getNextBuffer( temp );
+				const fpp_t frames = getNextBuffer( temp );
 				if( !frames )
 				{
 					quit = TRUE;
@@ -371,7 +371,7 @@ int audioALSA::setHWParams( const sample_rate_t _sample_rate,
 		}
 	}
 
-	m_periodSize = getMixer()->framesPerAudioBuffer();
+	m_periodSize = getMixer()->framesPerPeriod();
 	m_bufferSize = m_periodSize * 8;
 	dir = 0;
 	err = snd_pcm_hw_params_set_period_size_near( m_handle, m_hwParams,

src/audio/audio_device.cpp

@@ -42,7 +42,7 @@ audioDevice::audioDevice( const sample_rate_t _sample_rate,
 	m_sampleRate( _sample_rate ),
 	m_channels( _channels ),
 	m_mixer( _mixer ),
-	m_buffer( new surroundSampleFrame[getMixer()->framesPerAudioBuffer()] )
+	m_buffer( new surroundSampleFrame[getMixer()->framesPerPeriod()] )
 {
 	int error;
 	if( ( m_srcState = src_new(
@@ -81,7 +81,7 @@ audioDevice::~audioDevice()
 
 void audioDevice::processNextBuffer( void )
 {
-	const fpab_t frames = getNextBuffer( m_buffer );
+	const fpp_t frames = getNextBuffer( m_buffer );
 	if( frames )
 	{
 		writeBuffer( m_buffer, frames, getMixer()->masterGain() );
@@ -95,9 +95,9 @@ void audioDevice::processNextBuffer( void )
 
 
 
-fpab_t audioDevice::getNextBuffer( surroundSampleFrame * _ab )
+fpp_t audioDevice::getNextBuffer( surroundSampleFrame * _ab )
 {
-	fpab_t frames = getMixer()->framesPerAudioBuffer();
+	fpp_t frames = getMixer()->framesPerPeriod();
 	const surroundSampleFrame * b = getMixer()->nextBuffer();
 	if( !b )
 	{
@@ -163,7 +163,7 @@ void audioDevice::renamePort( audioPort * )
 
 
 void FASTCALL audioDevice::resample( const surroundSampleFrame * _src,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						surroundSampleFrame * _dst,
 						const sample_rate_t _src_sr,
 						const sample_rate_t _dst_sr )
@@ -190,7 +190,7 @@ void FASTCALL audioDevice::resample( const surroundSampleFrame * _src,
 
 
 Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const float _master_gain,
 						int_sample_t * _output_buffer,
 						const bool _convert_endian )
@@ -198,7 +198,7 @@ Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
 	if( _convert_endian )
 	{
 		Uint16 temp;
-		for( fpab_t frame = 0; frame < _frames; ++frame )
+		for( fpp_t frame = 0; frame < _frames; ++frame )
 		{
 			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 			{
@@ -215,7 +215,7 @@ Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
 	}
 	else
 	{
-		for( fpab_t frame = 0; frame < _frames; ++frame )
+		for( fpp_t frame = 0; frame < _frames; ++frame )
 		{
 			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 			{
@@ -235,7 +235,7 @@ Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
 
 
 void FASTCALL audioDevice::clearS16Buffer( int_sample_t * _outbuf,
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 #ifdef LMMS_DEBUG
 	assert( _outbuf != NULL );

src/audio/audio_file_ogg.cpp

@@ -179,7 +179,7 @@ bool audioFileOgg::startEncoding( void )
 
 
 void FASTCALL audioFileOgg::writeBuffer( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const float _master_gain )
 {
 	int eos = 0;
@@ -187,7 +187,7 @@ void FASTCALL audioFileOgg::writeBuffer( const surroundSampleFrame * _ab,
 	float * * buffer = vorbis_analysis_buffer( &m_vd, _frames *
 							BYTES_PER_SAMPLE *
 								channels() );
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
 		{

src/audio/audio_file_wave.cpp

@@ -92,7 +92,7 @@ bool audioFileWave::startEncoding( void )
 
 
 void FASTCALL audioFileWave::writeBuffer( const surroundSampleFrame * _ab,
-						const fpab_t _frames,
+						const fpp_t _frames,
 						const float _master_gain )
 {
 	int_sample_t * outbuf = new int_sample_t[_frames * channels()];

src/audio/audio_jack.cpp

@@ -67,7 +67,7 @@ audioJACK::audioJACK( const sample_rate_t _sample_rate, bool & _success_ful,
 	m_client( NULL ),
 	m_active( FALSE ),
 	m_stop_semaphore( 1 ),
-	m_outBuf( new surroundSampleFrame[getMixer()->framesPerAudioBuffer()] ),
+	m_outBuf( new surroundSampleFrame[getMixer()->framesPerPeriod()] ),
 	m_framesDoneInCurBuf( 0 ),
 	m_framesToDoInCurBuf( 0 )
 {
@@ -232,7 +232,7 @@ void audioJACK::startProcessing( void )
 
 
 	// try to sync JACK's and LMMS's buffer-size
-	jack_set_buffer_size( m_client, getMixer()->framesPerAudioBuffer() );
+	jack_set_buffer_size( m_client, getMixer()->framesPerPeriod() );
 
 
 
@@ -369,7 +369,7 @@ int audioJACK::processCallback( jack_nframes_t _nframes, void * _udata )
 	}
 
 /*	const Uint32 frames = tMin<Uint32>( _nframes,
-					getMixer()->framesPerAudioBuffer() );
+					getMixer()->framesPerPeriod() );
 	for( jackPortMap::iterator it = _this->m_portMap.begin();
 					it != _this->m_portMap.end(); ++it )
 	{

src/audio/audio_oss.cpp

@@ -127,7 +127,7 @@ audioOSS::audioOSS( const sample_rate_t _sample_rate, bool & _success_ful,
 
 	int frag_spec;
 	for( frag_spec = 0; static_cast<int>( 0x01 << frag_spec ) <
-		getMixer()->framesPerAudioBuffer() * channels() *
+		getMixer()->framesPerPeriod() * channels() *
 							BYTES_PER_INT_SAMPLE;
 		++frag_spec )
 	{
@@ -316,14 +316,14 @@ void audioOSS::stopProcessing( void )
 void audioOSS::run( void )
 {
 	surroundSampleFrame * temp =
-		new surroundSampleFrame[getMixer()->framesPerAudioBuffer()];
+		new surroundSampleFrame[getMixer()->framesPerPeriod()];
 	int_sample_t * outbuf =
-			new int_sample_t[getMixer()->framesPerAudioBuffer() *
+			new int_sample_t[getMixer()->framesPerPeriod() *
 								channels()];
 
 	while( TRUE )
 	{
-		const fpab_t frames = getNextBuffer( temp );
+		const fpp_t frames = getNextBuffer( temp );
 		if( !frames )
 		{
 			break;

src/audio/audio_port.cpp

@@ -36,19 +36,19 @@
 audioPort::audioPort( const QString & _name ) :
 	m_bufferUsage( NONE ),
 	m_firstBuffer( new surroundSampleFrame[
-				engine::getMixer()->framesPerAudioBuffer()] ),
+				engine::getMixer()->framesPerPeriod()] ),
 	m_secondBuffer( new surroundSampleFrame[
-				engine::getMixer()->framesPerAudioBuffer()] ),
+				engine::getMixer()->framesPerPeriod()] ),
 	m_extOutputEnabled( FALSE ),
 	m_nextFxChannel( -1 ),
 	m_name( "unnamed port" ),
 	m_effects( new effectChain ),
-	m_frames( engine::getMixer()->framesPerAudioBuffer() )
+	m_frames( engine::getMixer()->framesPerPeriod() )
 {
 	engine::getMixer()->clearAudioBuffer( m_firstBuffer,
-				engine::getMixer()->framesPerAudioBuffer() );
+				engine::getMixer()->framesPerPeriod() );
 	engine::getMixer()->clearAudioBuffer( m_secondBuffer,
-				engine::getMixer()->framesPerAudioBuffer() );
+				engine::getMixer()->framesPerPeriod() );
 	engine::getMixer()->addAudioPort( this );
 	setExtOutputEnabled( TRUE );
 }
@@ -71,7 +71,7 @@ audioPort::~audioPort()
 void audioPort::nextPeriod( void )
 {
 	engine::getMixer()->clearAudioBuffer( m_firstBuffer,
-				engine::getMixer()->framesPerAudioBuffer() );
+				engine::getMixer()->framesPerPeriod() );
 	qSwap( m_firstBuffer, m_secondBuffer );
 	// this is how we decrease state of buffer-usage ;-)
 	m_bufferUsage = ( m_bufferUsage != NONE ) ?

src/audio/audio_sample_recorder.cpp

@@ -102,10 +102,10 @@ void audioSampleRecorder::createSampleBuffer( sampleBuffer * * _sample_buf )
 
 
 void audioSampleRecorder::writeBuffer( const surroundSampleFrame * _ab,
-					const fpab_t _frames, const float )
+					const fpp_t _frames, const float )
 {
 	sampleFrame * buf = new sampleFrame[_frames];
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
 		{

src/audio/audio_sdl.cpp

@@ -54,14 +54,14 @@
 audioSDL::audioSDL( const sample_rate_t _sample_rate, bool & _success_ful,
 							mixer * _mixer ) :
 	audioDevice( _sample_rate, DEFAULT_CHANNELS, _mixer ),
-	m_outBuf( new surroundSampleFrame[getMixer()->framesPerAudioBuffer()] ),
+	m_outBuf( new surroundSampleFrame[getMixer()->framesPerPeriod()] ),
 	m_convertedBuf_pos( 0 ),
 	m_convertEndian( FALSE ),
 	m_stop_semaphore( 1 )
 {
 	_success_ful = FALSE;
 
-	m_convertedBuf_size = getMixer()->framesPerAudioBuffer() * channels()
+	m_convertedBuf_size = getMixer()->framesPerPeriod() * channels()
 						* sizeof( int_sample_t );
 	m_convertedBuf = new Uint8[m_convertedBuf_size];
 
@@ -90,7 +90,7 @@ audioSDL::audioSDL( const sample_rate_t _sample_rate, bool & _success_ful,
 						// of system, so we don't have
 						// to convert the buffers
 	m_audioHandle.channels = channels();
-	m_audioHandle.samples = getMixer()->framesPerAudioBuffer();
+	m_audioHandle.samples = getMixer()->framesPerPeriod();
 
 	m_audioHandle.callback = sdlAudioCallback;
 	m_audioHandle.userdata = this;
@@ -190,7 +190,7 @@ void audioSDL::sdlAudioCallback( Uint8 * _buf, int _len )
 		if( m_convertedBuf_pos == 0 )
 		{
 			// frames depend on the sample rate
-			const fpab_t frames = getNextBuffer( m_outBuf );
+			const fpp_t frames = getNextBuffer( m_outBuf );
 			if( !frames )
 			{
 				m_stopped = TRUE;

src/core/arp_and_chords_tab_widget.cpp

@@ -468,7 +468,7 @@ void arpAndChordsTabWidget::processNote( notePlayHandle * _n )
 				// create sub-note-play-handle, only note is
 				// different
 				new notePlayHandle( _n->getInstrumentTrack(),
-							_n->framesAhead(),
+							_n->offset(),
 							_n->frames(), note_copy,
 							_n );
 			}
@@ -524,13 +524,13 @@ void arpAndChordsTabWidget::processNote( notePlayHandle * _n )
 	// used for loop
 	f_cnt_t frames_processed = 0;
 
-	while( frames_processed < engine::getMixer()->framesPerAudioBuffer() )
+	while( frames_processed < engine::getMixer()->framesPerPeriod() )
 	{
 		const f_cnt_t remaining_frames_for_cur_arp = arp_frames -
 						( cur_frame % arp_frames );
 		// does current arp-note fill whole audio-buffer?
 		if( remaining_frames_for_cur_arp >
-				engine::getMixer()->framesPerAudioBuffer() )
+				engine::getMixer()->framesPerPeriod() )
 		{
 			// then we don't have to do something!
 			break;
@@ -622,8 +622,8 @@ void arpAndChordsTabWidget::processNote( notePlayHandle * _n )
 		// and is_arp_note=TRUE
 		new notePlayHandle( _n->getInstrumentTrack(),
 				( ( m_arpModeComboBox->value() != FREE ) ?
-						cnphv.first()->framesAhead() :
-						_n->framesAhead() ) +
+						cnphv.first()->offset() :
+						_n->offset() ) +
 							frames_processed,
 						gated_frames,
 						new_note,

src/core/bb_editor.cpp

@@ -227,8 +227,8 @@ tact bbEditor::lengthOfBB( csize _bb )
 
 
 
-bool FASTCALL bbEditor::play( midiTime _start, fpab_t _frames,
-							f_cnt_t _frame_base,
+bool FASTCALL bbEditor::play( midiTime _start, fpp_t _frames,
+							f_cnt_t _offset,
 							Sint16 _tco_num )
 {
 	bool played_a_note = FALSE;
@@ -242,7 +242,7 @@ bool FASTCALL bbEditor::play( midiTime _start, fpab_t _frames,
 	trackVector tv = tracks();
 	for( trackVector::iterator it = tv.begin(); it != tv.end(); ++it )
 	{
-		if( ( *it )->play( _start, _frames, _frame_base,
+		if( ( *it )->play( _start, _frames, _offset,
 							_tco_num ) == TRUE )
 		{
 			played_a_note = TRUE;

src/core/effect.cpp

@@ -67,7 +67,7 @@ effect::~effect()
 
 
 bool FASTCALL effect::processAudioBuffer( surroundSampleFrame * _buf, 
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	return( FALSE );
 }
@@ -78,7 +78,7 @@ bool FASTCALL effect::processAudioBuffer( surroundSampleFrame * _buf,
 void FASTCALL effect::setGate( float _level )
 {
 	m_gate = _level * _level * m_processors * 
-				engine::getMixer()->framesPerAudioBuffer();
+				engine::getMixer()->framesPerPeriod();
 }
 
 

src/core/effect_chain.cpp

@@ -117,7 +117,7 @@ void FASTCALL effectChain::moveUp( effect * _effect )
 
 
 bool FASTCALL effectChain::processAudioBuffer( surroundSampleFrame * _buf, 
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	if( m_bypassed )
 	{

src/core/envelope_and_lfo_widget.cpp

@@ -481,7 +481,7 @@ envelopeAndLFOWidget::envelopeAndLFOWidget( float _value_for_zero_amount,
 						SLOT( updateSampleVars() ) );
 
 	m_lfoShapeData =
-		new sample_t[engine::getMixer()->framesPerAudioBuffer()];
+		new sample_t[engine::getMixer()->framesPerPeriod()];
 	updateSampleVars();
 }
 
@@ -504,7 +504,7 @@ envelopeAndLFOWidget::~envelopeAndLFOWidget()
 
 
 
-inline sample_t envelopeAndLFOWidget::lfoShapeSample( fpab_t _frame_offset )
+inline sample_t envelopeAndLFOWidget::lfoShapeSample( fpp_t _frame_offset )
 {
 	f_cnt_t frame = ( m_lfoFrame + _frame_offset ) % m_lfoOscillationFrames;
 	const float phase = frame / static_cast<float>(
@@ -537,8 +537,8 @@ inline sample_t envelopeAndLFOWidget::lfoShapeSample( fpab_t _frame_offset )
 
 void envelopeAndLFOWidget::updateLFOShapeData( void )
 {
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
-	for( fpab_t offset = 0; offset < frames; ++offset )
+	const fpp_t frames = engine::getMixer()->framesPerPeriod();
+	for( fpp_t offset = 0; offset < frames; ++offset )
 	{
 		m_lfoShapeData[offset] = lfoShapeSample( offset );
 	}
@@ -555,7 +555,7 @@ void envelopeAndLFOWidget::triggerLFO( void )
 							it != v.end(); ++it )
 	{
 		( *it )->m_lfoFrame +=
-				engine::getMixer()->framesPerAudioBuffer();
+				engine::getMixer()->framesPerPeriod();
 		( *it )->m_bad_lfoShapeData = TRUE;
 	}
 }
@@ -579,11 +579,11 @@ void envelopeAndLFOWidget::resetLFO( void )
 
 inline void FASTCALL envelopeAndLFOWidget::fillLFOLevel( float * _buf,
 							f_cnt_t _frame,
-							const fpab_t _frames )
+							const fpp_t _frames )
 {
 	if( m_lfoAmountIsZero || _frame <= m_lfoPredelayFrames )
 	{
-		for( fpab_t offset = 0; offset < _frames; ++offset )
+		for( fpp_t offset = 0; offset < _frames; ++offset )
 		{
 			*_buf++ = 0.0f;
 		}
@@ -596,7 +596,7 @@ inline void FASTCALL envelopeAndLFOWidget::fillLFOLevel( float * _buf,
 		updateLFOShapeData();
 	}
 
-	fpab_t offset = 0;
+	fpp_t offset = 0;
 	for( ; offset < _frames && _frame < m_lfoAttackFrames; ++offset,
 								++_frame )
 	{
@@ -613,11 +613,11 @@ inline void FASTCALL envelopeAndLFOWidget::fillLFOLevel( float * _buf,
 
 void FASTCALL envelopeAndLFOWidget::fillLevel( float * _buf, f_cnt_t _frame,
 					const f_cnt_t _release_begin,
-					const fpab_t _frames )
+					const fpp_t _frames )
 {
 	fillLFOLevel( _buf, _frame, _frames );
 
-	for( fpab_t offset = 0; offset < _frames; ++offset, ++_buf, ++_frame )
+	for( fpp_t offset = 0; offset < _frames; ++offset, ++_buf, ++_frame )
 	{
 		float env_level;
 		if( _frame < _release_begin )

src/core/envelope_tab_widget.cpp

@@ -234,7 +234,7 @@ float FASTCALL envelopeTabWidget::volumeLevel( notePlayHandle * _n,
 
 	if( _n->released() == FALSE )
 	{
-		release_begin += engine::getMixer()->framesPerAudioBuffer();
+		release_begin += engine::getMixer()->framesPerPeriod();
 	}
 
 	float volume_level;
@@ -248,7 +248,7 @@ float FASTCALL envelopeTabWidget::volumeLevel( notePlayHandle * _n,
 
 
 void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
-							const fpab_t _frames,
+							const fpp_t _frames,
 							notePlayHandle * _n )
 {
 	const f_cnt_t total_frames = _n->totalFramesPlayed();
@@ -257,7 +257,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 
 	if( _n->released() == FALSE )
 	{
-		release_begin += engine::getMixer()->framesPerAudioBuffer();
+		release_begin += engine::getMixer()->framesPerPeriod();
 	}
 
 	// because of optimizations, there's special code for several cases:
@@ -299,7 +299,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 		if( m_envLFOWidgets[CUT]->used() &&
 			m_envLFOWidgets[RES]->used() )
 		{
-			for( fpab_t frame = 0; frame < _frames; ++frame )
+			for( fpp_t frame = 0; frame < _frames; ++frame )
 			{
 				float new_cut_val = envelopeAndLFOWidget::expKnobVal( cut_buf[frame] ) * CUT_FREQ_MULTIPLIER +
 						m_filterCutKnob->value();
@@ -323,7 +323,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 		}
 		else if( m_envLFOWidgets[CUT]->used() )
 		{
-			for( fpab_t frame = 0; frame < _frames; ++frame )
+			for( fpp_t frame = 0; frame < _frames; ++frame )
 			{
 				float new_cut_val = envelopeAndLFOWidget::expKnobVal( cut_buf[frame] ) * CUT_FREQ_MULTIPLIER +
 						m_filterCutKnob->value();
@@ -342,7 +342,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 		}
 		else if( m_envLFOWidgets[RES]->used() )
 		{
-			for( fpab_t frame = 0; frame < _frames; ++frame )
+			for( fpp_t frame = 0; frame < _frames; ++frame )
 			{
 				float new_res_val = m_filterResKnob->value() + RES_MULTIPLIER *
 							res_buf[frame];
@@ -363,7 +363,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 		{
 			_n->m_filter->calcFilterCoeffs( m_filterCutKnob->value(), m_filterResKnob->value() );
 
-			for( fpab_t frame = 0; frame < _frames; ++frame )
+			for( fpp_t frame = 0; frame < _frames; ++frame )
 			{
 				for( ch_cnt_t chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
 				{
@@ -382,7 +382,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 		m_envLFOWidgets[VOLUME]->fillLevel( vol_buf, total_frames,
 						release_begin, _frames );
 
-		for( fpab_t frame = 0; frame < _frames; ++frame )
+		for( fpp_t frame = 0; frame < _frames; ++frame )
 		{
 			float vol_level = vol_buf[frame];
 			vol_level = vol_level * vol_level;
@@ -398,7 +398,7 @@ void envelopeTabWidget::processAudioBuffer( sampleFrame * _ab,
 /*	else if( m_envLFOWidgets[VOLUME]->used() == FALSE && m_envLFOWidgets[PANNING]->used() )
 	{
 		// only use panning-envelope...
-		for( fpab_t frame = 0; frame < _frames; ++frame )
+		for( fpp_t frame = 0; frame < _frames; ++frame )
 		{
 			float vol_level = pan_buf[frame];
 			vol_level = vol_level*vol_level;

src/core/sample_play_handle.cpp

@@ -112,23 +112,15 @@ samplePlayHandle::~samplePlayHandle()
 
 
 
-void samplePlayHandle::play( bool _try_parallelizing )
-{
-	play( 0, _try_parallelizing );
-}
-
-
-
-
-void samplePlayHandle::play( const fpab_t _frame_base, bool )
+void samplePlayHandle::play( bool /* _try_parallelizing */ )
 {
+	//play( 0, _try_parallelizing );
 	if( framesDone() >= totalFrames() )
 	{
 		return;
 	}
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer()
-								- _frame_base;
+	const fpp_t frames = engine::getMixer()->framesPerPeriod();
 	if( !( m_track && m_track->muted() )
 				&& !( m_bbTrack && m_bbTrack->muted() ) )
 	{
@@ -139,7 +131,7 @@ void samplePlayHandle::play( const fpab_t _frame_base, bool )
 #endif
 				} } ;
 		m_sampleBuffer->play( buf, &m_state, frames );
-		engine::getMixer()->bufferToPort( buf, frames, _frame_base, v,
+		engine::getMixer()->bufferToPort( buf, frames, offset(), v,
 								m_audioPort );
 
 		delete[] buf;

src/core/song_editor.cpp

@@ -990,10 +990,10 @@ void songEditor::processNextBuffer( void )
 	float frames_per_tact64th = engine::framesPerTact64th();
 
 	while( total_frames_played
-				< engine::getMixer()->framesPerAudioBuffer() )
+				< engine::getMixer()->framesPerPeriod() )
 	{
 		f_cnt_t played_frames = engine::getMixer()
-				->framesPerAudioBuffer() - total_frames_played;
+				->framesPerPeriod() - total_frames_played;
 
 		float current_frame = m_playPos[m_playMode].currentFrame();
 		// did we play a 64th of a tact?

src/lib/oscillator.cpp

@@ -52,7 +52,7 @@ oscillator::oscillator( const waveShapes & _wave_shape,
 
 
 
-void oscillator::update( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::update( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	if( m_subOsc != NULL )
@@ -84,7 +84,7 @@ void oscillator::update( sampleFrame * _ab, const fpab_t _frames,
 
 
 
-void oscillator::updateNoSub( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateNoSub( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	switch( m_waveShape )
@@ -120,7 +120,7 @@ void oscillator::updateNoSub( sampleFrame * _ab, const fpab_t _frames,
 
 
 
-void oscillator::updatePM( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updatePM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	switch( m_waveShape )
@@ -156,7 +156,7 @@ void oscillator::updatePM( sampleFrame * _ab, const fpab_t _frames,
 
 
 
-void oscillator::updateAM( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateAM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	switch( m_waveShape )
@@ -192,7 +192,7 @@ void oscillator::updateAM( sampleFrame * _ab, const fpab_t _frames,
 
 
 
-void oscillator::updateMix( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateMix( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	switch( m_waveShape )
@@ -228,7 +228,7 @@ void oscillator::updateMix( sampleFrame * _ab, const fpab_t _frames,
 
 
 
-void oscillator::updateSync( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateSync( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	switch( m_waveShape )
@@ -264,7 +264,7 @@ void oscillator::updateSync( sampleFrame * _ab, const fpab_t _frames,
 
 
 
-void oscillator::updateFM( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateFM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	switch( m_waveShape )
@@ -326,7 +326,7 @@ inline bool oscillator::syncOk( float _osc_coeff )
 
 
 
-float oscillator::syncInit( sampleFrame * _ab, const fpab_t _frames,
+float oscillator::syncInit( sampleFrame * _ab, const fpp_t _frames,
 						const ch_cnt_t _chnl )
 {
 	if( m_subOsc != NULL )
@@ -342,13 +342,13 @@ float oscillator::syncInit( sampleFrame * _ab, const fpab_t _frames,
 
 // if we have no sub-osc, we can't do any modulation... just get our samples
 template<oscillator::waveShapes W>
-void oscillator::updateNoSub( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateNoSub( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	recalcPhase();
 	const float osc_coeff = m_freq * m_detuning;
 
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		_ab[frame][_chnl] = getSample<W>( m_phase ) * m_volume;
 		m_phase += osc_coeff;
@@ -360,14 +360,14 @@ void oscillator::updateNoSub( sampleFrame * _ab, const fpab_t _frames,
 
 // do pm by using sub-osc as modulator
 template<oscillator::waveShapes W>
-void oscillator::updatePM( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updatePM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	m_subOsc->update( _ab, _frames, _chnl );
 	recalcPhase();
 	const float osc_coeff = m_freq * m_detuning;
 
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		_ab[frame][_chnl] = getSample<W>( m_phase + _ab[frame][_chnl] )
 								* m_volume;
@@ -380,14 +380,14 @@ void oscillator::updatePM( sampleFrame * _ab, const fpab_t _frames,
 
 // do am by using sub-osc as modulator
 template<oscillator::waveShapes W>
-void oscillator::updateAM( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateAM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	m_subOsc->update( _ab, _frames, _chnl );
 	recalcPhase();
 	const float osc_coeff = m_freq * m_detuning;
 
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		_ab[frame][_chnl] *= getSample<W>( m_phase ) * m_volume;
 		m_phase += osc_coeff;
@@ -399,14 +399,14 @@ void oscillator::updateAM( sampleFrame * _ab, const fpab_t _frames,
 
 // do mix by using sub-osc as mix-sample
 template<oscillator::waveShapes W>
-void oscillator::updateMix( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateMix( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	m_subOsc->update( _ab, _frames, _chnl );
 	recalcPhase();
 	const float osc_coeff = m_freq * m_detuning;
 
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		_ab[frame][_chnl] += getSample<W>( m_phase ) * m_volume;
 		m_phase += osc_coeff;
@@ -419,14 +419,14 @@ void oscillator::updateMix( sampleFrame * _ab, const fpab_t _frames,
 // sync with sub-osc (every time sub-osc starts new period, we also start new
 // period)
 template<oscillator::waveShapes W>
-void oscillator::updateSync( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateSync( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	const float sub_osc_coeff = m_subOsc->syncInit( _ab, _frames, _chnl );
 	recalcPhase();
 	const float osc_coeff = m_freq * m_detuning;
 
-	for( fpab_t frame = 0; frame < _frames ; ++frame )
+	for( fpp_t frame = 0; frame < _frames ; ++frame )
 	{
 		if( m_subOsc->syncOk( sub_osc_coeff ) )
 		{
@@ -442,14 +442,14 @@ void oscillator::updateSync( sampleFrame * _ab, const fpab_t _frames,
 
 // do fm by using sub-osc as modulator
 template<oscillator::waveShapes W>
-void oscillator::updateFM( sampleFrame * _ab, const fpab_t _frames,
+void oscillator::updateFM( sampleFrame * _ab, const fpp_t _frames,
 							const ch_cnt_t _chnl )
 {
 	m_subOsc->update( _ab, _frames, _chnl );
 	recalcPhase();
 	const float osc_coeff = m_freq * m_detuning;
 
-	for( fpab_t frame = 0; frame < _frames; ++frame )
+	for( fpp_t frame = 0; frame < _frames; ++frame )
 	{
 		m_phase += _ab[frame][_chnl];
 		_ab[frame][_chnl] = getSample<W>( m_phase ) * m_volume;

src/lib/sample_buffer.cpp

@@ -594,7 +594,7 @@ void sampleBuffer::initResampling( void )
 
 
 bool FASTCALL sampleBuffer::play( sampleFrame * _ab, handleState * _state,
-					const fpab_t _frames,
+					const fpp_t _frames,
 					const float _freq,
 					const bool _looped )
 {

src/tracks/automation_track.cpp

@@ -55,7 +55,7 @@ track::trackTypes automationTrack::type( void ) const
 
 
 bool automationTrack::play( const midiTime & _start,
-					const fpab_t _frames,
+					const fpp_t _frames,
 					const f_cnt_t _frame_base,
 							Sint16 _tco_num )
 {

src/tracks/bb_track.cpp

@@ -404,8 +404,8 @@ track::trackTypes bbTrack::type( void ) const
 
 // play _frames frames of given TCO within starting with _start
 bool FASTCALL bbTrack::play( const midiTime & _start,
-						const fpab_t _frames,
-						const f_cnt_t _frame_base,
+						const fpp_t _frames,
+						const f_cnt_t _offset,
 							Sint16 _tco_num )
 {
 	sendMidiTime( _start );
@@ -413,7 +413,7 @@ bool FASTCALL bbTrack::play( const midiTime & _start,
 	if( _tco_num >= 0 )
 	{
 		return( engine::getBBEditor()->play( _start, _frames,
-							_frame_base,
+							_offset,
 							s_infoMap[this] ) );
 	}
 
@@ -442,7 +442,7 @@ bool FASTCALL bbTrack::play( const midiTime & _start,
 	{
 		return( engine::getBBEditor()->play( _start - lastPosition,
 							_frames,
-							_frame_base,
+							_offset,
 							s_infoMap[this] ) );
 	}
 	return( FALSE );

src/widgets/rack_plugin.cpp

@@ -279,7 +279,7 @@ void rackPlugin::setAutoQuit( float _value )
 {
 	float samples = engine::getMixer()->sampleRate() * _value / 1000.0f;
 	Uint32 buffers = 1 + ( static_cast<Uint32>( samples ) / 
-			engine::getMixer()->framesPerAudioBuffer() );
+			engine::getMixer()->framesPerPeriod() );
 	m_effect->setTimeout( buffers );
 }
 

src/widgets/visualization_widget.cpp

@@ -66,7 +66,7 @@ visualizationWidget::visualizationWidget( const QPixmap & _bg, QWidget * _p,
 	setFixedSize( s_background.width(), s_background.height() );
 
 
-	const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
+	const fpp_t frames = engine::getMixer()->framesPerPeriod();
 	m_buffer = new surroundSampleFrame[frames];
 
 	engine::getMixer()->clearAudioBuffer( m_buffer, frames );
@@ -139,11 +139,11 @@ void visualizationWidget::paintEvent( QPaintEvent * )
 
 		float max_level = 0.0;
 
-		const fpab_t frames =
-				engine::getMixer()->framesPerAudioBuffer();
+		const fpp_t frames =
+				engine::getMixer()->framesPerPeriod();
 
 		// analyse wave-stream for max-level
-		for( fpab_t frame = 0; frame < frames; ++frame )
+		for( fpp_t frame = 0; frame < frames; ++frame )
 		{
 			for( ch_cnt_t chnl = 0; chnl < SURROUND_CHANNELS;
 									++chnl )
@@ -171,7 +171,7 @@ void visualizationWidget::paintEvent( QPaintEvent * )
 		}
 
 		// now draw all that stuff
-		for( fpab_t frame = 0; frame < frames; ++frame )
+		for( fpp_t frame = 0; frame < frames; ++frame )
 		{
 			for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
 			{