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

include/audio_alsa.h

@@ -1,7 +1,7 @@
 /*
  * audio_alsa.h - device-class that implements ALSA-PCM-output
  *
- * 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
  *
@@ -51,8 +51,7 @@ class QLineEdit;
 class audioALSA : public audioDevice, public QThread
 {
 public:
-	audioALSA( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer );
+	audioALSA( bool & _success_ful, mixer * _mixer );
 	virtual ~audioALSA();
 
 	inline static QString name( void )
@@ -84,11 +83,9 @@ private:
 	virtual void stopProcessing( void );
 	virtual void run( void );
 
-	int FASTCALL setHWParams( const sample_rate_t _sample_rate,
-						const ch_cnt_t _channels,
-						snd_pcm_access_t _access );
+	int setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access );
 	int setSWParams( void );
-	int FASTCALL handleError( int _err );
+	int handleError( int _err );
 
 
 	snd_pcm_t * m_handle;

include/audio_device.h

@@ -1,7 +1,7 @@
 /*
  * audio_device.h - 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
  *
@@ -34,13 +34,6 @@
 #include <config.h>
 #endif
 
-#ifndef USE_3RDPARTY_LIBSRC
-#include <samplerate.h>
-#else
-#include "src/3rdparty/samplerate/samplerate.h"
-#endif
-
-
 #include "mixer.h"
 #include "tab_widget.h"
 
@@ -53,8 +46,7 @@ class audioPort;
 class audioDevice
 {
 public:
-	audioDevice( const sample_rate_t _sample_rate,
-				const ch_cnt_t _channels, mixer * _mixer );
+	audioDevice( const ch_cnt_t _channels, mixer * _mixer );
 	virtual ~audioDevice();
 
 	inline void lock( void )
@@ -128,29 +120,29 @@ public:
 protected:
 	// subclasses can re-implement this for being used in conjunction with
 	// processNextBuffer()
-	virtual void FASTCALL writeBuffer( const surroundSampleFrame * _ab,
+	virtual void writeBuffer( const surroundSampleFrame * _ab,
 						const fpp_t _frames,
 						const float _master_gain )
 	{
 	}
 
 	// called by according driver for fetching new sound-data
-	fpp_t FASTCALL getNextBuffer( surroundSampleFrame * _ab );
+	fpp_t 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,
+	Uint32 convertToS16( const surroundSampleFrame * _ab,
 					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,
+	void clearS16Buffer( int_sample_t * _outbuf,
 							const fpp_t _frames );
 
 	// resample given buffer from samplerate _src_sr to samplerate _dst_sr
-	void FASTCALL resample( const surroundSampleFrame * _src,
+	void resample( const surroundSampleFrame * _src,
 					const fpp_t _frames,
 					surroundSampleFrame * _dst,
 					const sample_rate_t _src_sr,

include/audio_dummy.h

@@ -1,7 +1,7 @@
 /*
  * audio_dummy.h - dummy-audio-device
  *
- * 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
  *
@@ -33,9 +33,8 @@
 class audioDummy : public audioDevice, public QThread
 {
 public:
-	audioDummy( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer ) :
-		audioDevice( _sample_rate, DEFAULT_CHANNELS, _mixer )
+	audioDummy( bool & _success_ful, mixer * _mixer ) :
+		audioDevice( DEFAULT_CHANNELS, _mixer )
 	{
 		_success_ful = TRUE;
 	}
@@ -48,7 +47,7 @@ public:
 	inline static QString name( void )
 	{
 		return( QT_TRANSLATE_NOOP( "setupWidget",
-			"Dummy (no sound output)" ) );
+						"Dummy (no sound output)" ) );
 	}
 
 
@@ -110,7 +109,8 @@ private:
 
 			const Sint32 microseconds = static_cast<Sint32>(
 					getMixer()->framesPerPeriod() *
-					1000000.0f / getMixer()->sampleRate() -
+					1000000.0f /
+				getMixer()->processingSampleRate() -
 							timer.elapsed() );
 			if( microseconds > 0 )
 			{

include/audio_file_device.h

@@ -2,7 +2,7 @@
  * audio_file_device.h - base-class for audio-device-classes which write 
  *                       their output into a file
  *
- * Copyright (c) 2004-2006 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
  *

include/audio_jack.h

@@ -1,7 +1,7 @@
 /*
  * audio_jack.h - support for JACK-transport
  *
- * Copyright (c) 2005-2007 Tobias Doerffel <tobydox/at/users.sourceforge.net>
+ * Copyright (c) 2005-2008 Tobias Doerffel <tobydox/at/users.sourceforge.net>
  * 
  * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
  *
@@ -52,8 +52,7 @@ class lcdSpinBox;
 class audioJACK : public audioDevice
 {
 public:
-	audioJACK( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer );
+	audioJACK( bool & _success_ful, mixer * _mixer );
 	virtual ~audioJACK();
 
 	inline static QString name( void )

include/audio_oss.h

@@ -1,7 +1,7 @@
 /*
  * audio_oss.h - device-class that implements OSS-PCM-output
  *
- * 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
  *
@@ -46,8 +46,7 @@ class QLineEdit;
 class audioOSS : public audioDevice, public QThread
 {
 public:
-	audioOSS( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer );
+	audioOSS( bool & _success_ful, mixer * _mixer );
 	virtual ~audioOSS();
 
 	inline static QString name( void )

include/audio_sample_recorder.h

@@ -3,7 +3,7 @@
  *                           surround-audio-buffers into RAM, maybe later
  *                           also harddisk
  *
- * Copyright (c) 2004-2006 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
  *
@@ -41,17 +41,16 @@ class sampleBuffer;
 class audioSampleRecorder : public audioDevice
 {
 public:
-	audioSampleRecorder( const sample_rate_t _sample_rate,
-				const ch_cnt_t _channels, bool & _success_ful,
+	audioSampleRecorder( const ch_cnt_t _channels, bool & _success_ful,
 							mixer * _mixer );
 	virtual ~audioSampleRecorder();
 
 	f_cnt_t framesRecorded( void ) const;
-	void FASTCALL createSampleBuffer( sampleBuffer * * _sample_buf );
+	void createSampleBuffer( sampleBuffer * * _sample_buf );
 
 
 private:
-	virtual void FASTCALL writeBuffer( const surroundSampleFrame * _ab,
+	virtual void writeBuffer( const surroundSampleFrame * _ab,
 						const fpp_t _frames,
 						const float _master_gain );
 

include/audio_sdl.h

@@ -1,7 +1,7 @@
 /*
  * audio_sdl.h - 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
  *
@@ -50,14 +50,13 @@ class QLineEdit;
 class audioSDL : public audioDevice
 {
 public:
-	audioSDL( const sample_rate_t _sample_rate, bool & _success_ful,
-							mixer * _mixer );
+	audioSDL( bool & _success_ful, mixer * _mixer );
 	virtual ~audioSDL();
 
 	inline static QString name( void )
 	{
 		return( QT_TRANSLATE_NOOP( "setupWidget",
-			"SDL (Simple DirectMedia Layer)" ) );
+					"SDL (Simple DirectMedia Layer)" ) );
 	}
 
 

include/controller.h

@@ -52,7 +52,9 @@ public:
 
 	inline bool isSampleExact( void ) const
 	{
-		return m_sampleExact || engine::getMixer()->highQuality();
+		return m_sampleExact ||
+			engine::getMixer()->qualitySettings().
+							sampleExactControllers;
 	}
 
 	void setSampleExact( bool _exact )

include/effect.h

@@ -109,7 +109,8 @@ public:
 
 	inline f_cnt_t getTimeout( void ) const
 	{
-		const float samples = engine::getMixer()->sampleRate() *
+		const float samples =
+				engine::getMixer()->processingSampleRate() *
 					m_autoQuitModel.value() / 1000.0f;
 		return( 1 + ( static_cast<Uint32>( samples ) / 
 				engine::getMixer()->framesPerPeriod() ) );

include/mixer.h

@@ -30,6 +30,13 @@
 #include <config.h>
 #endif
 
+#ifndef USE_3RDPARTY_LIBSRC
+#include <samplerate.h>
+#else
+#include "src/3rdparty/samplerate/samplerate.h"
+#endif
+
+
 #include <QtCore/QMutex>
 #include <QtCore/QSemaphore>
 #include <QtCore/QThread>
@@ -58,16 +65,6 @@ const ch_cnt_t SURROUND_CHANNELS =
 #endif
 
 
-enum qualityLevels
-{
-	DEFAULT_QUALITY_LEVEL,
-	HIGH_QUALITY_LEVEL,
-	QUALITY_LEVELS
-} ;
-
-extern sample_rate_t SAMPLE_RATES[QUALITY_LEVELS];
-const sample_rate_t DEFAULT_SAMPLE_RATE = 44100;
-
 
 typedef sample_t sampleFrame[DEFAULT_CHANNELS];
 typedef sample_t surroundSampleFrame[SURROUND_CHANNELS];
@@ -96,8 +93,103 @@ class mixer : public QObject
 {
 	Q_OBJECT
 public:
+	struct qualitySettings
+	{
+		enum Mode
+		{
+			Mode_Draft,
+			Mode_HighQuality,
+			Mode_FinalMix
+		} ;
+		enum Interpolation
+		{
+			Interpolation_Linear,
+			Interpolation_SincFastest,
+			Interpolation_SincMedium,
+			Interpolation_SincBest
+		} ; 
+
+		enum Oversampling
+		{
+			Oversampling_None,
+			Oversampling_2x,
+			Oversampling_4x,
+			Oversampling_8x,
+			Oversampling_16x
+		} ;
+
+		Interpolation interpolation;
+		Oversampling oversampling;
+		bool sampleExactControllers;
+		bool aliasFreeOscillators;
+
+		qualitySettings( Mode _m )
+		{
+			switch( _m )
+			{
+				case Mode_Draft:
+					interpolation = Interpolation_Linear;
+					oversampling = Oversampling_None;
+					sampleExactControllers = FALSE;
+					aliasFreeOscillators = FALSE;
+					break;
+				case Mode_HighQuality:
+					interpolation =
+						Interpolation_SincFastest;
+					oversampling = Oversampling_2x;
+					sampleExactControllers = TRUE;
+					aliasFreeOscillators = FALSE;
+					break;
+				case Mode_FinalMix:
+					interpolation = Interpolation_SincBest;
+					oversampling = Oversampling_8x;
+					sampleExactControllers = TRUE;
+					aliasFreeOscillators = TRUE;
+					break;
+			}
+		}
+
+		qualitySettings( Interpolation _i, Oversampling _o, bool _sec,
+								bool _afo ) :
+			interpolation( _i ),
+			oversampling( _o ),
+			sampleExactControllers( _sec ),
+			aliasFreeOscillators( _afo )
+		{
+		}
+
+		int sampleRateMultiplier( void ) const
+		{
+			switch( oversampling )
+			{
+				case Oversampling_None: return 1;
+				case Oversampling_2x: return 2;
+				case Oversampling_4x: return 4;
+				case Oversampling_8x: return 8;
+				case Oversampling_16x: return 16;
+			}
+			return( 1 );
+		}
+
+		int libsrcInterpolation( void ) const
+		{
+			switch( interpolation )
+			{
+				case Interpolation_Linear:
+					return( SRC_LINEAR );
+				case Interpolation_SincFastest:
+					return( SRC_SINC_FASTEST );
+				case Interpolation_SincMedium:
+					return( SRC_SINC_MEDIUM_QUALITY );
+				case Interpolation_SincBest:
+					return( SRC_SINC_BEST_QUALITY );
+			}
+			return( SRC_LINEAR );
+		}
+	} ;
+
 	void initDevices( void );
-	void FASTCALL clear( void );
+	void clear( void );
 
 
 	// audio-device-stuff
@@ -106,7 +198,7 @@ public:
 		return( m_audioDevName );
 	}
 
-	void FASTCALL setAudioDevice( audioDevice * _dev, bool _hq );
+	void setAudioDevice( audioDevice * _dev );
 	void restoreAudioDevice( void );
 	inline audioDevice * audioDev( void )
 	{
@@ -199,16 +291,15 @@ public:
 		return( m_cpuLoad );
 	}
 
-	inline bool highQuality( void ) const
+	const qualitySettings & qualitySettings( void ) const
 	{
-		return( m_qualityLevel > DEFAULT_QUALITY_LEVEL );
+		return( m_qualitySettings );
 	}
 
 
-	inline sample_rate_t sampleRate( void ) const
-	{
-		return( SAMPLE_RATES[m_qualityLevel] );
-	}
+	sample_rate_t baseSampleRate( void ) const;
+	sample_rate_t outputSampleRate( void ) const;
+	sample_rate_t processingSampleRate( void ) const;
 
 
 	inline float masterGain( void ) const
@@ -268,7 +359,7 @@ public:
 	}
 
 	// audio-buffer-mgm
-	void FASTCALL bufferToPort( const sampleFrame * _buf,
+	void bufferToPort( const sampleFrame * _buf,
 					const fpp_t _frames,
 					const f_cnt_t _offset,
 					stereoVolumeVector _volume_vector,
@@ -295,11 +386,11 @@ public:
 	}
 
 
-public slots:
-	void setHighQuality( bool _hq_on = FALSE );
+	void changeQuality( const struct qualitySettings & _qs );
 
 
 signals:
+	void qualitySettingsChanged( void );
 	void sampleRateChanged( void );
 	void nextAudioBuffer( void );
 
@@ -371,7 +462,7 @@ private:
 	playHandleVector m_playHandles;
 	constPlayHandleVector m_playHandlesToRemove;
 
-	qualityLevels m_qualityLevel;
+	struct qualitySettings m_qualitySettings;
 	float m_masterGain;
 
 

include/sample_buffer.h

@@ -80,13 +80,12 @@ public:
 	
 	virtual ~sampleBuffer();
 
-	bool FASTCALL play( sampleFrame * _ab, handleState * _state,
+	bool play( sampleFrame * _ab, handleState * _state,
 				const fpp_t _frames,
 				const float _freq = BaseFreq,
 				const bool _looped = FALSE ) const;
 
-	void FASTCALL visualize( QPainter & _p, const QRect & _dr,
-							const QRect & _clip );
+	void visualize( QPainter & _p, const QRect & _dr, const QRect & _clip );
 	inline void visualize( QPainter & _p, const QRect & _dr )
 	{
 		visualize( _p, _dr, _dr );
@@ -157,13 +156,12 @@ public:
 	QString & toBase64( QString & _dst ) const;
 
 
-	static sampleBuffer * FASTCALL resample( sampleFrame * _data,
+	static sampleBuffer * resample( sampleFrame * _data,
 						const f_cnt_t _frames,
 						const sample_rate_t _src_sr,
 						const sample_rate_t _dst_sr );
 
-	static inline sampleBuffer * FASTCALL resample(
-						sampleBuffer * _buf,
+	static inline sampleBuffer * resample( sampleBuffer * _buf,
 						const sample_rate_t _src_sr,
 						const sample_rate_t _dst_sr )
 	{
@@ -171,7 +169,7 @@ public:
 								_dst_sr ) );
 	}
 
-	void normalize_sample_rate( const sample_rate_t _src_sr,
+	void normalizeSampleRate( const sample_rate_t _src_sr,
 						bool _keep_settings = FALSE );
 
 	inline sample_t userWaveSample( const float _sample ) const
@@ -209,29 +207,25 @@ public slots:
 
 
 private:
-	void FASTCALL update( bool _keep_settings = FALSE );
+	void update( bool _keep_settings = FALSE );
 
 
 #ifdef SDL_SDL_SOUND_H
-	f_cnt_t FASTCALL decodeSampleSDL( const char * _f,
-						int_sample_t * & _buf,
+	f_cnt_t decodeSampleSDL( const char * _f, int_sample_t * & _buf,
 						ch_cnt_t _channels,
 						sample_rate_t _sample_rate );
 #endif
 #ifdef HAVE_SNDFILE_H
-	f_cnt_t FASTCALL decodeSampleSF( const char * _f,
-						int_sample_t * & _buf,
+	f_cnt_t decodeSampleSF( const char * _f, int_sample_t * & _buf,
 						ch_cnt_t & _channels,
 						sample_rate_t & _sample_rate );
 #endif
 #ifdef HAVE_VORBIS_VORBISFILE_H
-	f_cnt_t FASTCALL decodeSampleOGGVorbis( const char * _f,
-						int_sample_t * & _buf,
+	f_cnt_t decodeSampleOGGVorbis( const char * _f, int_sample_t * & _buf,
 						ch_cnt_t & _channels,
 						sample_rate_t & _sample_rate );
 #endif
-	f_cnt_t FASTCALL decodeSampleDS( const char * _f,
-						int_sample_t * & _buf,
+	f_cnt_t decodeSampleDS( const char * _f, int_sample_t * & _buf,
 						ch_cnt_t & _channels,
 						sample_rate_t & _sample_rate );
 

include/song.h

@@ -95,24 +95,24 @@ public:
 	void processNextBuffer( void );
 
 
-	inline bool paused( void ) const
+	inline bool isPaused( void ) const
 	{
 		return( m_paused );
 	}
 
-	inline bool playing( void ) const
+	inline bool isPlaying( void ) const
 	{
 		return( m_playing && m_exporting == FALSE );
 	}
 
-	inline bool exporting( void ) const
+	inline bool isExporting( void ) const
 	{
 		return( m_exporting );
 	}
 
 	bool realTimeTask( void ) const;
 
-	inline bool exportDone( void ) const
+	inline bool isExportDone( void ) const
 	{
 		return( m_exporting == TRUE &&
 			m_playPos[Mode_PlaySong].getTact() >= length() + 1 );
@@ -194,7 +194,7 @@ public slots:
 	void exportProject( void );
 
 	void startExport( void );
-	void stopExport( void );
+	void cancelExport( void );
 
 
 	void setModified( void );